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f41th Issue 14
yyyyyssssyyyy yyyyssssyyyy yyyy yyyy
|lS$$ yy $$$$ """" yy lS$$ S$$$ S$$$$$ $$$$$ S$$$ssssyyyy
:|lS$ ""yyyyy yyyyssss|lS$ lS$$ lS$$ yy$$$$$ lS$$ yy lS$$
:||lS$$ $$$$$ :|lS yy :|lS |lS$ |lS$ $$ yyyy |lS$ $$ |lS$
:::|l ,$$$$$ ::|l $$ ::|l :|lS :|lS $$ :|lS :|lS $$ :|lS
::::| $$$$$$ :::| $$ :::| ::|l ::|l $$ ::|l ::|l $$ ::|l
.:::: ....... .:::....:::: .::| ..:|....:::| .::| .. .::|
=========================================================
F41th 14. May 2001. http://www.f41th.com. - D4RKCYDE -
=========================================================
.
:
|
+----> Message of the 0day hybrid
+----> Mailbag various
+----> Digital Multiplexing System janus
+----> Meridian 1 PBX Admin: Part II Sneeka
+----> Telewest VoIP re-load
+----> Modem Brown-Box phractal
+----> Time-Division Multiplexing foneman
+----> UK Trunking Network Primer hybrid
|
:
.
.
:
|
+--> Message of the 0day <--+
|
:
+--> hybrid <hybrid@f41th.com>
The last f41th (f41th 13) was released in June 2000.. thats nearly a whole
year ago now.. Why has everything been so idle? --dont ask me, maybee its
somthing to do with lack of interest in phones and/or this mag. Some people
sent me some highly crap "articles" for the zine, (probably over 15 of them),
however, the ^M's and bad formating from lame cut and paste techniqs annoyed me,
so I have left them out. I do not see the point in publishing somthing in this
zine if it was directly cut'n'pasted from a reference guide, and especially if
it was formated in windows..
When D4RKCYDE first started f41th, we recieved some good articles from people...
stop sending shit, we have plently of docs allready, we dont need your cut'n'pasted
windows pdf shit.
-hybrid
.
:
|
+--> Mailbag <--+
|
:
+--> *@f41th.com
Date: Wed, 24 Jan 2001 13:20:18 -0000
From: sue denim <@hotmail.com>
To: hybrid@f41th.com
Subject: request for help!
whatsup?
--> WAAAAAAAZZZZUUUUUUUUPPPPPP
wasn't sure who to write to about this;
basically i'm a hacker/phreaker newbie; looking for some info on:-
manipulating an ericsson T28s? hiding my user identity/ making free calls?
--> nice phone.
what modern PAYT mobile is easiest to manipulate, and cheap to obtain?
cracking email passwords??!
--> people send you passwds in their emails?
how can access the software on a simcard and change it?
im using netscape 4.0, what kind of 'mischeif' can i get up to?
--> wtf are you talking about??
basically i'm a complete newbie in modern computer software, but keen to
learn unix/linux? whats a good starting point?
--> http://linux.org | .com
what can i do from a public shared machine? i'd love to hack/crack the fuck
out of it!!
--> yeah, you do that..
can i hack/crack from a laptop using a cellphone?
--> yes
send me some info; i'd be stoked and super grateful!!
cheers.
Date: Sat, 10 Feb 2001 11:11:14 -0700
From: no no <@hotmail.com>
To: hybrid@f41th.com
Subject: hi i wanna.....
Hi i wanna join the uk phreak scene, i was wondering if u could maybe
"mentor" me in my quest for ultimate knowledge :)If u are in any sort of
phreak/hack team then maybe u could introduce me to a few people and my
learning would come along a lot quicker,
Thanks for yur time in writing those texts..
Da_0ne
--> i am not in the uk phreak scene, the uk scene + D4RKCYDE are seperate
--> things.. if you want to join the "scene" read alt.ph.uk, its elite,
--> you will lean many things there. if you are looking to be a phreak,
--> just stay away from the "scene" its pants, you'll realise that after
--> a while... do your own thing.. come talk in #darkcyde efnet if you
--> want to meet other ppl.
Date: 14 Feb 2001 14:51:51 -0000
From: nasir xyz <@rediffmail.com>
To: hybrid@f41th.com <hybrid@f41th.com>
Subject: outgoing calls
Dear friend,
I found yr articles highly intersting I from country India .
I know that you people can help me fooling my exchange .
PLs tell me how can i make free outgoing call telephone calls from my home
telephone(for example my exchange think tha my
telephone is still ringing and i am not connected to the dail number) ,Is it possible or not,
Can i do it by changing polarity with the help of diodes or Any other way.
Thanks in advance/
Nasirxyz
--> hmm, interesting idea about tricking the exchange into thinking the dialed number is
--> still ringing.. i dont know how to do this though, probably not even possible depending
--> on the types of exhanges involved.. if you exchange is digital, i suspect that this is
--> impossible because of signalling reasons... dont' know though, might be worth looking into.
Date: Wed, 14 Feb 2001 16:17:59 -0500
From: joel <@home.com>
To: hybrid@f41th.com
Subject: hey
hey there .you wouldn't happen to know of any phoe number tracing software. i have a
small company and we get all kinds
of prank calls. I got caller id and they just come up no info. They are probley
hitting *67 like i do. Any help you could
give would be great....
JD
-->
Hi there. Thanks for the info. I called the telco Co. and they are coming
out next week to trap my line they said. Hopefully that helps. A few pranks
were funny but the way they talk to the women that work here is uncalled
for. Thanks again very much...Jd
--> NP!
Date: Sat, 17 Feb 2001 02:27:01 +0800
From: nein wan wan <@spam.org>
To: hybrid@f41th.com
Subject: Your publication
I am interested in getting involved with your publication. Is this possible? You have
probably talked to me before, as
I've been around for a while. I'm an old fart wanting to get back into the scene.
Your response would be greatly
appreciated. -- nein --
--> sure, cometo #darkcyde on efnet and we'll talk about it.
Date: Thu, 22 Feb 2001 10:39:11 -0800 (PST)
From: John Wilson <@yahoo.com>
To: hybrid@f41th.com
Subject: Dear Hybrid:
Hi, My Name Is John.I'm 17 Years Old And I Have Been
Reading Your Articles About The "Underground
Telecommunications". If You Would Please Fill Me In A
Little More About This Subject.I'm Highly Intrested
And I Got All The Time It Takes. The So Called
"Government" Is Hideing Something From You And I And I
Will Die To Find Out What That Is. I Would Appreciate
You Mailing Me Back. Thank You Vey Much..
Sincerely,
John
--> http://hybrid.dtmf.org ..
Date: Thu, 22 Feb 2001 15:22:03 -0600 (CST)
From: @hushmail.com
To: hybrid@f41th.com
Subject: info
Was wondering if you guys had an updated list on hackers under investigation.
The fbi has been investigating my group and they just took some of my computer
shit, so we kinda want to know heh
--> what kind of shit did they take?
yea...my dedicated server.
--> D4RKCYDE has been idle for a while now, we havn't done anything to draw
--> much attention of that type to us.. I remember when we where "under investigation"
--> from the FBI (from antionline), probably bullshit though, never paid attention to
--> that..
Date: Thu, 22 Feb 2001 21:24:31 -0000
From: A Osman <@btinternet.com>
To: hybrid@f41th.com
Subject: Pay as you go
Hi
I hope you and your friends could shed some light on this situation.
I have been collecting used One2One pay as you go vouchers in a vain attempt
to try and crack the code that they use. Dont know if you deal with mobiles
but any help would be great.
Wokky
--> One2One probably have some extreamly random procedure for generating the
--> codes, i wouldnt waste my time if i where you..
Date: 7 Mar 2001 15:19:17 -0800
From: Atlantic Lab <@techemail.com>
To: hybrid@f41th.com
Subject: Necromancy!
I'm just a small potato in a large potato field but I want to grow.
How would I do so!
Where would I start?
--> you got me..
Date: 8 Mar 2001 13:55:35 -0800
From: Atlantic Lab <@techemail.com>
To: hybrid@f41th.com
Subject: more magic please!
The Bode' Darma sat in front of a wall in a cave for 9
years. When he felt that he was finished, he left and the world became a richer place
because of his teachings, Men found a
harmony and richness within themselves through physical and spiritual expression!
--> please stop emailing me, you are scary.
Date: Thu, 8 Mar 2001 22:41:24 -0800 (PST)
From: dogg <@yahoo.com>
To: hybrid@f41th.com
Subject: information request
I'm not really sure I have the right email address, but i was wondering...is there a way
to actually find a persons email
code with a cd program. please respond if you have any information on any related
topics. thanks
--> email code?
Date: Fri, 09 Mar 2001 09:18:58 -0000
From: Sharpish Me <@hotmail.com>
To: hybrid@f41th.com
Subject: F41th mirror...
I read the f41th txt's at college, found them interesting, so I mirror'd
them on my w/s. I'm a newb to all this, I just find it interesting...
the link:
http://www.simpletown.com/sharpish/f41th/f41thmenu.htm
There ya go.
Sharpish
++++++++++++++++++
"Wake me when the future happens..."
++++++++++++++++++
--> thanks, i'll put it on the site links if you update it.
Date: 10 Mar 2001 10:56:09 -0800
From: Atlantic Lab <@techemail.com>
To: hybrid@f41th.com
Subject: last magic!
It seems to me that whatever understanding we share of each other is going to be our personal undoing!
I assure you I'm just another individual with tremendous interest in what you have written, not who you are!
There is a project that I'm currently working on that will allow me to find data that doesn't want to be
found! All I ask is
possibly some feedback!
Ideas!
Input!
This is all!!!
If you choose not to correspond I will understand, and discontinue these transmissions!
Thank You :)
--> PLEASE STOP EMAILING ME!$£@$!
Date: Sat, 10 Mar 2001 19:13:50 +0000
From: @one2one.net
To: hybrid@F41th.com
Subject: Phone Phreaking !
Hi,
Is it possible to obtain free phone calls through phone phreaking ?
Thanks
Chris
--> yeah i guess so.
Date: Sun, 18 Mar 2001 16:54:20 +0000 (GMT)
From: "[iso-8859-1] bob bob" <@yahoo.co.uk>
To: hybrid@dtmf.org
hi,
i am a newbie phreak(thought it best to say before i
carry on and not be a bigger lamer by pretending i m
eliet or anything....) i have done a few things
however, i got in to some blokes vmb,listened to his
messages and changed his OGM. pretty small fry realy
but good for a start. anyways, the problem i have is
barred numbers on payphones(thats what ive done
everything through up till now) i can only dial 0800
no's, i wanted to know any methods of getting around
this, and also how numbers are barred? and ways of
baring/unbarring them. some nice posibilities there...
so anyway, keep up the good work on the d4rkcyde and
help me on the way to leaving lamerdom,
from
pa_p_ercllps
--> you changed someones OGM?? why did you do that yo? train
--> yourself to prowl in the shadows of the trunk, its the
--> way of the ninja.
Date: Mon, 26 Mar 2001 11:07:29 +0500
From: BEE ANN <@hotmail.com>
To: hybrid@f41th.com
Subject: C L I
Hi,
I read your articals, I am really inpired, I need your help in regard to stop my out going CLI,
I live in Pakistan, and out PTT dose not provide such services like CLI with held or CLIR
can you help me by telling me simple methods by using my pstn phone and stop my out-going CLI,
as a metter of fact we
know that some organisation have the ANI, but I want this for my personal use, our PTT has
two swithing systems 1)
Alcatel 2) Siemens.
Moreover, If you can help me by telling me the simple procedure to stop the out going CLI by
using my GSM 900 Mhz phone ,
the service provider of which use the Motorlla GSM switching systems.
I am dyeing to read your positive reply.
Thanking you in advance,
Bee
--> i'll look into it, i dont have much info on pakistan.
Date: Tue, 27 Mar 2001 14:12:24 +0200
From: Nirmal D Morjaria <@lycos.com>
To: hybrid@b4b0.org
Subject: (No Subject)
What is the purpose of these telephone numbers I have found on the internet.
In this text file are they to make free phone calls??
--> no dude, they are scans.. either hand-scans (looking for interesting shit) or
--> carrier scans (looking for modem carriers)
Date: Mon, 2 Apr 2001 13:49:15 -0700
From: <@msn.com>
To: hybrid@f41th.com
Subject: HELP
PLEASE HELP ME! I WANT TO KNOW IF MY GIRL IS CHEATING ON ME ,WOULD YOU BE ABLE TO TELL ME HOW TO HACK HER
VOICEMAIL?
--> DAMN$! its ok, sorry, i'll leave her alone, she told me she was single.
Date: Wed, 4 Apr 2001 04:57:04 EDT
From: @cs.com
To: hybrid@f41th.com
Subject: hey about your zine
hey I was wonderin if your guys magazine is actually published or if its
electronic, if its published is there anyway i can get a subscription in the
US? I hope so it sounds really bad ass, i just wanted to ask and pay my
respects to you guys so i hopefully will talk to one of you later, please
email me with an answer.
SiruX
Of The Dark Alliance
--> we have been thinking about publishing f41th as hard copy for a while now..
--> although we cannot find a printer, let alone somehwere to distribute..
--> maybee if people start sending good enough articles, we'll start printing..
--> for thew time being it will reamin in digital format.
Date: Sat, 7 Apr 2001 00:40:38 EDT
From: @cs.com
To: hybrid@f41th.com
Subject: Re: about your zine
cool, thank you for actually returning the email, the underrgound needs more
allies like you guys, aieght, well if you guys are excepting any articles or
anything like that reply to me, and if you want ill write for ya, aieght, see
ya!
SiRuX
--> werd dude, keep it real.
Date: Wed, 18 Apr 2001 02:16:12 +0100 (BST)
From: "[iso-8859-1] Martin Jones" <@yahoo.co.uk>
To: hybrid@f41th.com
Whosoever used the w32.baftrans.13312@mm virus and
without due reason deleted the Yahoo account of a
certain bluegrass playing duo is hereby given notice
that they are on Doctor Zargs target list and will
soon be suffering the retribution of the Fr33n3t 0rg.
Be aware: hacking is a noble and artistic past time
when directed against corporate entities and even
personal webpresences but when exacted in a
mischievous manner, resulting in destructive activity
against independent entities it is considered an act
of war.
As a result of this act the 4th & 5th echelons of
Albion Net Venturers are in Antagonist D mode.
You script kiddies can do no harm to the Fr33', but
many rising innocents are vulnerable and do not
deserve your ignorant vandalism.
Be aware we actively encourage and continuously
monitor anarchical activity across the web but cannot
condone desecration of the innocents.
I am sending this from my personal address as a mark
of my sincere wish to communicate my honest feelings
on this matter and do so in good faith without blame
against you or your community per say, but please
understand that this recent abuse has resulted in a
fully negative reaction fom the Freenet tech commune.
Long live the new flesh.
--> i have the fear.
Date: Fri, 20 Apr 2001 12:58:59 -0700 (PDT)
From: Webmaster <@b-w-d.net>
To: hybrid@f41th.com
Subject: Call-Waiting4free (BT got a ittle problem)
####################################
## Call-Waiting (Why pay for it?) ##
####################################
Im quite new to phreaking, but ive found an exploit
in the BT system allowing ANYONE to get call-waiting
on their line, free. Its not the complete service
(as you cant flick between calls with the BT menu)
but you can still keep a call on hold and phone
another number, the return to the original person.
Hey, u may think call waitin is crap, but it can be usefull, trust me =]
What you need:
===============
1. Quite an old phone
2. This text
Note: the phone MUST be able to use pulse dialing...and directly!
Preperation:
==============
To make sure your phone will let you do this, pick up the phone and hold down the
1 button, if it sends out a continuous
pulse, then you can do it, if it sends out seperate pulses, then ur fuckd. :P
[for all u simple ppl, what i mean is, when u press down the button, if the sound
coming from your phone is just a continueous
sound, then its ok, otherwise, if its like, click-click-click-click, your fuckd]
What ya do
===========
1. pick up the phone and call someone(the person your gonna put on hold, person 1).
2. Check your phone is set to pulse (if its capable of tone aswell).
3. Hold down the 1 button untill you hear a blank tone.
4. Make your next call (person 2).
5. Put down the phone when youve finished talking to person 2.
# The phone will now ring asif you askd a number to call back when its no longer busy #
6. Pick up the phone and your back with person 1.
7. Hang up as per usual.
The-Finger@BT.com
==================
hey, have u ever wonderd why u sometimes get rung back by people? well, thats coz you accidentaly
hit the receiver long enough
to send this pulse down the line. BT's FAULT!
yeah, BT does know about this, but its not fixed yet, and it usually only works from the person
who called, but sometimes it
works for both sides.
##############################################
Credz go to me (.-=LeGiOn=-.)
Vir0n (4 bein around 2 help)
Enclave-Guardian (for my big phone bill)
Every1-els@Brotherhood of Enclave
##############################################
enclave-hackers.com www.b-w-d.co.uk/hack/
##############################################
Contact me at @b-w-d.net ... if ya want proof of this being my work, i got the document
from BT about it, and it will
be e-maild/snailmaild on request.
.
:
|
+--> Digital Multiplexing System <--+
|
:
+--> janus
Digital Multiplexing System
===========================
--This file will attempt to explain the DMS (Digital Multiplexing System).
Think of this file as more of a compilation of the material I have read,
rather than something I authored completely from scratch. Special thanks to
Control-C for most of the information found here.
-DMS
====
DMS was/is made by Northern Telecom. It was first introduced in 1979. To
date, DMS has been able to interface with such switches as ESS #1-4, Xbar,
TSPS, and EAX. The DMS switch itself is physically smaller than a Xbar
switch, and usually smaller than most AXE switches. This is because the DMS
switch is more spread out, as opposed to other types of switches which are
all located in one switch house. The use of remote modules give the CO more
space to install a Line Concentrating Module (LCM) or Main Distribution
Frame (MDF). Many versions of DMS exist. DMS versions and systems are as
follows:
1) DMS-10 -
===========
a C5 switch which can be used with up to 10,800 lines. Designed
for rural areas and large businesses. Almost always connected with a larger
DMS-100 or -100/200 switch.
2) DMS-100 -
============
a C5 local office able to be used with 1,000 to 100,000 lines.
Very widely used today to handle residential areas' phone lines. A DMS-100
local office can also be adapted to Equal Access End Office (EAEO)
3) DMS-200 -
============
can be used with up to 60,000 trunks. Can also serve a AT
(Access Tandem) function. The Auxiliary Operator Services System (AOSS)
is a part of DMS-200 that controls Operater-assisted calls, such as
Directory Assistance. AOSS is made possible by Traffic Operator Position
System (TOPS) and Operator Centralization (OC). These 2 functions allow
transfer operator services from other DMS-200 toll centers.
4) DMS 100/200 -
================
Uses functions such as the toll and local systems mentioned
above, but also includes the EAEO/AT combination. Can handle either 100,000
lines or 60,000 trunks. Used instead of using -100 and -200 seperately.
5) DMS-250 -
============
Not very widely used. Used in association with specialized
common carriers that need tandem switching.
6) DMS-300 -
============
Designed for international use. The number of DMS-300 switches
that are used is in the single digits.
7) Remote Switching Center (RSC) -
==================================
Used instead of DMS-100, it has the
ability to switch up to 5,760 lines.
8) Remote Line Concentrating Module (RLCM) -
============================================
Able to switch up to 640 lines.
Can be used with RSC or DMS-100 with assistance from the Line Concentrator
Module (LCM).
9) Outside Plant Module (OPM) -
===============================
Able to switch up to 640 lines. Can also be
used in association with RSC or DMS-100.
10) Subscriber Carrier Module (SCM or SCM-100) -
================================================
-a) Subscriber Carrier Module (Rural (SCM-100R)) - Eliminates the CO
Central Control Terminal (CCT) by being integrated with a DMS-100 switch.
-b) Subscriber Carrier Module SLC-96 (SCM-100S) - gives a direct link
between DMS-100 and SLC-96 loop carriers.
-c) Subscriber Carrier Module Urban (SCM-100U) - Used to interact with
DMS-1 Urban (DMS version specialized for use in urban areas.)
11) DMS-Mobile Telephone Exhange (DMS-MTX) -
============================================
A special type of DMS-100 that is used with Cellular switching.
is used with Cellular switching. It can serve up to 50,000 people in up to
50 cells.
12) Supernode
=============
-a) DMS-Supernode - Revision of the DMS-100 that supports faster processing.
-b) DMS-Supernode SE - same as above, except in a reduced physical size, and
uses the Link Peripheral Processor (LPP).
Important Features of DMS-100:
==============================
1) Automatic Route Selection - automatically detects the best trunk for
routing toll and LD calls.
2) Station Message Detail Recording - an enhanced call logging system,keeps
track of times, dates, duration, etc.
3) Direct Inward System Access (DISA) - allows maintenance and
administration from remote terminals.
Operator Features included with DMS-200 and -100/200:
=====================================================
1) Traffic Operator Position System (TOPS) - gives certain functions to
handle incoming and outgoing calls.
2) Operator Centralization (OC) - Lets an operator interface with the switch
equipment itself. Allows calls to be routed from a remote DMS switch to a
host.
DMS is divided into 4 areas that each handle special operations:
================================================================
1) Central Control Complex (CCC) - Controls the functions that are used in
the other 3 areas. The CCC contains 4 units:
-a) Central Processing Unit: Each DMS switch contains 2 CPUs. The CPUs have
access to memory banks where stored programs and network data are located.
Consider the CPUs the "engines" of the switch. They process all incoming
data from outside lines.
-b) Program Store Memory Module: Associated with one CPU to contain the
program instructions needed to run programs on the switch. The second PS
contains duplicate instructions.
-c) Data Store Memory Module: Contains information such as customer
information and office data. The second DS is a duplicate that is used with
the second CPU.
-d) Central Message Controller: Controls the messages between the other
areas of the CCC and the Network Message Controller (NMC) in the various
Network Modules or the I/O controller. Both CPUs have access to the CMC.
2) Network (NET) - Network Modules handle the vocal aspect between the
Peripheral Modules and the Central Control Complex (CCC).
3) Peripheral Modules (PM) - Interface between analog trunks, subscriber
lines, and digital carrier spans (DS-1). Responsible for creating dialtones,
sending/receiving signalling, and checking the network.
Before 1984, the following types of PMs existed:
================================================
-a) Trunk Module - Changes speech into digital format to be sent through the
line. The TM also handles MF tones, test circuit announcement trunks, etc.
-b) Digital Carrier Module - gives a digital interface between the DMS
switch and the DS-1 digital carrier. The DS-1 signal consists of 24 voice
channels.
-c) Line Module - gives an interface for a maximum of 640 analog lines and
condenses the voice and signaling into two, three, or four DS-30,
32-channel speech links.
-d) Remote Line Module - same as above, except it controls the DMS switch
remotely. Can be used up to 150 miles away.
Since 1984, 10 more types were added:
=====================================
-a) Digital Trunk Controller - Interfaces up to 20 DS-1 lines, then sends
the DS-1 lines to the network.
-b) Line Group Controller - Can interface up to 20 DS-30 lines, and can
serve RSCs, RLCMs, or OPMs.
-c) Line Trunk Controller - has the ability to give interfaces to a maximum
of 20 outside ports from DS-30A speech links or DS-1 links to 16 network
side DS-30 speech links.
-d) Line Concentrating Module - An expanded version of the LTC, it can serve
up to 640 subscriber lines interfaced with 2-6 DS-30 speech links.
-e) Remote Switching Center - interfaces subscriber lines at a remote
location to a DMS-100 host. The RSC consists of the Line Concentrator
Module, Remote Cluster Controller, Remote Trunking, Remote-off-Remote, and
Emergency Stand-alone.
-f) Remote Line Concentrating Module - an LCM used from a remote location
from the DMS-100 host. Can handle up to 640 lines, sometimes used as
replacement for PBXs.
-g) Outside Plant Module - Outside plant remote unit. Handles 640 lines over
6 DS-1 Links.
-h) Subscriber Carrier Module - Remote interface for remote concentrators.
-i) SCM-100R - Can interface up to five DMS-1R Terminals. Each terminal can
handle up to 256 lines.
-j) SCM-100U - Can interface up to three DMS-1 Urban RTs. Each RT can
interface up to 576 POTS or special service lines.
4) Maintenance and Adminstration -
==================================
DMS provides different ways to maintain
and administrate the network. M&A is divided into 4 major groups:
-a) Administrative: Provides for the interrogation, collection and
modification of data.
-b) Internal Maintenance: Includes all DMS hardware (to the MDF) and
software.
-c) External Maintenance: Includes circuits on the transmission facility.
-d) Reporting: Include I/O facilities and the alarm system.
Common Channel Interoffice Signalling (CCIS) uses a dedicated line to
transmit data between offices, trunks, or trunk groups. CCIS-6 uses the
International Consultative Committee on Telephone and Telegraph (CCITT) No.
6 international standard. CCIS-7 added the ability to use CCIS with almost
all common DMS versions such as DMS-100, -200, -100/200, and -100/200 with
TOPS. CCIS-6 uses 2 types of Serving Offices (SO):
1) CCIS-BS:
===========
used for trunk signalling between COs. Transmits data such as numbers dialed,
numbers dialed from, and other routing information.
CCIS-BS put an end to Blue Boxing.
2) CCIS-DS:
===========
enables the use of touch-tone menu administration, such as voicemail, calling
card imput, and so forth.
Access Tandems:
===============
1) Equal Access (EA) gives a connection between Local Access and
Transport Areas (LATA). It provides such services as ANI, Automatic Message
Accounting (AMA) for both originating and terminating calls, and operator
service signalling.
2) Equal Office End Office (EAEO) gives a connection between interLATA
carriers and international carriers' POP.
3) Access Tandem with Equal Access End Office gives a connection from a
trunk tandem to ICs/INCs POP inside a LATA. It uses a two-stage
"overlap output pulsing" method which makes dialing quicker and
easier. The first stage identifies the INC dialed and picks a
reliable outgoing trunk. A connection is established from the INC
to the EAEO through the access tandem. The second stage processes
ANI and makes a connection to the called number through your specific
DMS switch type.
4) Access Tandem with a Non-Equal End Office uses Feature Group A, B, or
C to connect to an IC/INC. It uses standard Central Automatic Message Accouting
(CAMA) to palce a call through AT. Other services provided with DMS switches
used in urban areas are as follows:
1) Auxiliary Operator Services System (AOSS) -
==============================================
used primarily for directory assistance, and the intercept needs not included with
TOPS.
2) Integrated Business Network (IBN) -
======================================
commercial concept designed for business to have a small, private PBX. IBN can
be installed into a business to a Centrex Control Office or a Centrex Costumer
Unit with minor hardware adjustments. Features of IBN include the ability to
handle 30,000 lines, customer call records, centralized attendant
maintenance, administration functions, and direct inward dialing.
3) Electronic Switched Network (ESN) -
======================================
designed to meet needs of multi-location complexes. Used with SL-1 or -100 Digital
Business Communications Systems with networking features or a DMS-100 IBN host.
4) Specialized Common Carrier Service (SCCS) -
==============================================
provides conversion of analog and digital signals. Must be used with older analog
and digital signals. Must be used with older analog lines, sometimes also used with
newer digital lines.
DMS-MTX is a DMS switch used for switching radio and cellular signals.
DMS switches provide 3 basic types of cell switching:
1) Stand-alone switching is used by a MTX which is interfaced with one
or more C5 EOs with DID trunks. MTX is used with urban areas,
MTXC
====
for suburban areas, and MTXM for rural areas.
2) Combined switching is the most cost-effective type of MTX and is easy
to install. It can be incorporated into a DMS-100 switch and used
with cellular software.
3) Remote switching is accomplished by the Remote Switching Center (RSC)
alongside a Cell Site Controller (CSC). A Remote or Stand-alone
switch hosts the remote switch. Remote switching is not used in
urban areas.
__________________
Suggested Reading:
Understanding DMS; Control-C; 1987 (Most of my information came from
here!)
DMS Family of Digital Switching Systems; Erudite; ????
DMS-100; Jester Sluggo; ????
DMS-100 Family System; Northern Telecom; 1978
--Janus
janusx0r@hotmail.com
.
:
|
+--> Meridian 1 PBX Admin: Part II <--+
|
:
+--> Sneeka
Meridian 1 PBX Admin: Part II Typed up and slightly edited by Sneeka
=-=-=-=-=-=-=-=-=-=-=-=-=-=-= (sneeka1@hotmail.com)
Ok, here we go, the second part. If you read the first part you should know
all about how the basic system operates and know how to move and change
extensions etc.
This part will concentrate on the various features you can apply to extensions
along with guides for implementing them.
Sneeka
--------------------------------------------------------------------------------
FEATURE ACTIVATION CODES
========================
Introduction
------------
Features are activated and deactivated by keying codes into the telephone
set and, in the case of digital sets, by pressing single keys for certain
features. There are two methods of invoking features, namely Special Prefix
(SPRE) code working and Flexible Feature Code (FFC) working.
Flexible Feature Code (FFC) Working
-----------------------------------
Flexible Feature Code working allows the sysadmin to assign their own codes to
chosen features. This allows each Meridian 1 to be tailored to a logical
pattern of codes to suit each company or to retain the same feature codes as
those used on their previous telephone system. FFCs are set and printed
through Program 57.
The Option 11 default FFCs are given in the table below:
Feature FFC
----------------------------------
Ring Again #31
Cancel Ring Again #32
Answer Call Pickup **0
Answer Group Pickup *00
Directed (DN) Pickup *09
Authorisation Code *21
Call Park #51
Retrieve Parked Call #52
Dial System Speed Call *3
Store Speed Call #33
Dial Speed Call #34
Call Forward all Calls #1
Hold (Permanent) #55
Store Number for Redial *24
Dial Stored Number Redial *25
Last Number Redial *44
Group Hunt Opt In #58
Group Hunt Opt Out #59
Automatic Set Relocation *90
Automatic Set Removal *92
Night Service Pickup *8
Remote Call Forward *56
Override *26
Charge Account #56
SPRE Code Working
-----------------
This method of working is not very common. These codes are used by extension
users without multi-frequency tone key phones. [ in other words, pulse
dialling
- Snk ]
SPRE code working consist of two parts:
SPRE code followed by FFC
=========================
The SPRE code tells the system that a feature is about to be activated or
deactivated and the FFC identifies the particular feature.
The SPRE code can be changed to be any value (max. 4 digits) so long as it
doesn't clash with the extension number range or access codes for routes.
The Option 11 system default SPRE code is 12.
The FFCs however cannot be changed. Common codes are below:
Feature FFC
----------------------------------
Ring Again 1
Cancel Ring Again 2
Answer Call Pickup 3
Answer Group Pickup 94
Directed (DN) Pickup 95
Authorisation Code 6
Call Park 71
Retrieve Parked Call 72
Dial System Speed Call 73
Store Speed Call 75
Dial Speed Call 76
Call Forward all Calls 74
Hold (Permanent) 77
Store Number for Redial 78
Dial Stored Number Redial 79
Last Number Redial 89
Night Service Pickup 4
Charge Account 5
Automatic Set Relocation 81
Activating Features From Digital Sets
-------------------------------------
With digital sets some features must be assigned to a programmable key
whereas other features can be invoked either by dialling a feature code
or be assigned to a key.
The following features if used must be assigned to a programmable key for
activation:
o Call Forward All Calls
o Ring Again
o Stored Number Redial
o Last Number Redial
o Speed Call User and Controller
o System Speed Call Controller
o Six Party Conference
o Call Transfer
o Call Park (Activation)
o Call Waiting
The following features may be assigned to a key or invoked using the relevant
feature code:
o Call Park (retrieve)
o System Speed Call User
o Call Pickup
o Charge Account
Listing all the FFCs on the System
==================================
You can print out the current code for all FFCs on the system using Program
57.
>LD 57
REQ PRT
TYPE FFC
CUST 0
CODE ALL
The system responds by printing all FFCs, and a typical printout is shown
below.
CUST 00
FFCT YES
ASRC *90
AUTH *21
AWUA
AWUD
AWUV
CDRC #56
CFWA #1
CFWD #1
CFWV
COND
CPAC #52
CPRK #51
C6DS
DEAF
DPVS
ELKA
ELKD
PLDN 2000
USE GPHT
LSNO 050
HTYP RRB
CFWI NO
MQUE 0
2000
|
|
|
|
HOLD #55
ICPA
ICPD
ICPO
ICPP
IMS
MNTC *93
MTRC
OVRD *26
PUDN *09
PUGR *00
PURN **0
RCFA *56
RCFD *57
RCFV
RDLN *44
RDNE
RDSN *25
RDST *24
RGAA #31
RGAD #32
RGAV
RMST
SCPC
SPCC #33
SPCE #36
SPCU #34
TFAS *8
TRMD
TRVS *95
USTA
LILO
NRDY
GHTA #58
GHTD #59
SADS
SAEN
SALK
SAUN
EOVR
AREM *92
ADMN *91
Listing a Particular FFC
========================
You can identify the code for a particular feature by using the Program 57.
The following example shows how to print the FCC associated with the "Ring
Again Activation (RGAA)" feature.
>LD 57
REQ PRT
TYPE FFC
CUST 0
CODE RGAA
The system responds with:
CUST 00
FFCT YES
RGAA #31
CODE {Hit Return here to get back to the REQ prompt}
REQ END {Exit}
Adding a New FFC
================
A code for any FFC can be added using Program 57.
The following example shows a new "Ring Again Activation Code (RGAA)" of #66
being added.
>LD 57
REQ CHG {Meaning 'Change' - used even if you're adding a new code}
TYPE FFC
CUST 0
FFCT
CODE RGAA {The feature mnemonic}
RGAA #66 {The code number}
RGAA {Return through this if you don't want to define any more
features}
CODE {Return here too}
REQ END {Exit}
Deleting a FCC
==============
Again, use Program 57 to delete an FFC.
The following shows a "Ring Again Activation Code (RGAA)" of #31 being
deleted:
>LD 57
REQ OUT
TYPE FFC
CUST O
ALL NO
CODE RGAA
RGAA #31
RGAA {Carriage Return to confirm the delete}
CODE {Return to indicate no other features are to be deleted}
REQ END {Exit}
Changing a FFC
==============
To change an FFC just delete the existing code and then add the new code.
FEATURE PACKAGING
=================
Introduction
------------
Most features are supplied as standard with the system. Optional feature
groups can be bought for specific applications eg. Automatic Call Distribution
(ACD). This section demonstrates how to list the features on a system and shows
how to identify the software package containing any particular feature.
Printing the Features that Exist on the System
==============================================
You can list the features currently on the system by loading Program 22 and
following the steps below.
>LD 22
REQ PRT
TYPE PKG {Meaning 'Packages'}
The system responds by listing the mnemonic for all the feature packages
installed on the system.
OPTF
CUST
CDR
CTY
CLNK
RAN
TAD
.
.
.
SR
AA
HIST
BARS
NARS
XPE
XCT0
XCT1
NACD
REQ END {Exit}
FEATURE DESCRIPTIONS
====================
Autodial
========
Description
-----------
The Autodial feature allows extension users with digital sets to store a
telephone number under an "Autodial Key" and when desired make a call to
that number by pressing the key. The sysadmin can restrict the length
of number that an extension user can store under the key to one of the following
4, 8, 12, 16, 20 or 23. To store the number under an Autodial press the key, then
key in the number and finally press the Autodial key again to save the number.
To use the autodial key first press an idle DN key and then press the Autodial key.
Implementation
--------------
To configure this feature first identify the TN of the extension, the key to
be used and then determine the maximum number of digits the extension user will
be allowed to store. Then you can either input a stored number or leave
it for the extension user to input from their telephone. Use Program 11 as
follows:
ITEM KEY xx ADL yy zzzzzzzz
where xx = key number
yy = maximum length of Autodial number
zz = the digits to be dialled automatically (optional)
Example
-------
In this example the Autodial feature has been assigned to key 8 of a digital
set located at TN 8 3. The user is allowed to store up to 16 digits and the
number 908658081 is initially stored.
>LD 11
REQ CHG
TYPE 2616
TN 8 3 {Identify the TN}
ECHG YES {Use EasyChange}
ITEM KEY 8 ADL 16 908658081 {Allocate Autodial to key 8, limit the
number of
digits stored to 16 and store 908658081}
ITEM {No more keys to configure so Return through
this}
REQ END {Exit}
Automatic Answerback
====================
Description
-----------
This feature is useful to anyone that requires complete hands free operation
of their M2616 type set. When the feature is activated any incoming calls to
that set will ring the phone once and then be automatically answered.
Implementation
--------------
This feature can be implemented in two ways.
The first is by setting the Class of Service to AAA. This will cause every
incoming call to ring the extension once and then be automatically answered.
The second is to allocate the feature to a key. This allows the extension
user to activate or deactivate the feature as desired.
Example
-------
In this example the Automatic Answerback feature has been assigned to key 8
of a digital set located at TN 8 3.
>LD 11
REQ CHG
TYPE 2616
TN 8 3
ECHG YES
ITEM KEY 8 AAK
KEY
ITEM
REQ END {Exit}
Automatic Hold
==============
Description
-----------
This feature can only be assigned to digital sets. It allows users to place
a call on hold without using the "Hold" key. For sets with multiple DNs the
user can have two or more calls active at the same time and switch between the
calls without placing the active call on hold. As the DN of a held call is
pressed to retrieve that call the active call is automatically put on hold.
Attempting to switch between DNs without this feature would cause the active
call to be cut off.
Implementation
--------------
This feature is configured in the Class of Service in Program 11.
Example
-------
In this example the Automatic Hold feature has been assigned to a digital
set located at TN 8 3.
>LD 11
REQ CHG
TYPE 2616
TN 8 3
ECHG YES
ITEM CLS AHA
ITEM
REQ END {Exit}
Authorisation Codes
===================
Description
-----------
This feature gives "authorised personnel" a mechanism to temporarily
override the Call Restriction (barring) applicable to an extension thus allowing them
to make a from normally barred extensions. Such an example might be a manager
who is regularly called to other parts of the building. The manager need only
dial the Authorisation Code FFC (*21 is the default) followed by their "Personal
Identification Number (PIN)" and, when dial tone is returned, the number
they require. This action will then override any barring for one call.
Implementation
--------------
Authorisation codes are added, changed, deleted and printed in Program 88.
They are programmed in two parts. First to define the authorisation codes,
and secondly to define the call restrictions applied to that code:
a) Type AUB (Authorisation data block). This allows different combinations
of Class of Service (COS), Trunk Group Access Rescriction (TGAR) and/or
Network Class of Service (NCOS) to be defined. The different combinations of
COS TGAR and NCOS are assigned to what is termed a classcode (CLAS). There
can be a maximum of 116 (0 to 115) different CLAS on the system.
AUB
----------------------------------
CLAS COS
----------------------------------
0 UNR, TGAR=1
1 TLD, NCOS=2
2 UNR
.
.
115 UNR
b) Type AUT (Authorisation code Entries). This is where the individual PIN
numbers are defined. The Authorisation Code (CODE), or PIN, is assigned
to a Classcode value (CLAS) which will determine any restriction, that will
apply when the code is used.
AUT
---------------------
CODE CLAS
---------------------
2745 0
2632 1
9789 2
3428 1
6679 2
Example
-------
In this example two new Authorisation Codes with unrestricted will be set
up. First print the AUB to find a CLAS which has a COS = UNR (unrestricted).
Then print the AUT to identify which CODEs (PINs) are already in use. Then
create two new CODEs:
>LD 88
REQ PRT
TYPE AUB
CUST 0
The system responds by printing the Authorisation Data Block. From this you
can see that CLAS 1 has a COS of UNR.
CUST 00
ALEN 04
ACDR YES
RANR X
AUTO NO
CLAS COS TGAR NCOS CLAS COS TGAR NCOS CLAS COS TGAR NCOS
--------------------- --------------------- ---------------------
000 TLD 01 07 001 UNR 00 00 002 TLD 00 02
003 UNR 00 00 004 UNR 00 00 005 UNR 00 00
006 UNR 00 00 007 UNR 00 00 008 UNR 00 00
009 UNR 00 00 010 UNR 00 00 011 UNR 00 00
012 UNR 00 00 013 UNR 00 00 014 UNR 00 00
. . . . . .
. . . . . .
111 UNR 00 00 112 UNR 00 00 113 UNR 00 00
114 UNR 00 00 115 UNR 00 00
To find the authorisation codes are in use.
REQ PRT
TYPE AUT
CUST 0
CODE
The system responds with the list.
CODE 1111 CLAS 001
CODE 1234 CLAS 000
Now two new codes can be set up, eg. 5218 and 2465.
REQ NEW
TYPE AUT
CUST 0
CODE 5218
SARC {Return at this leaves the prompt at default. The default
is
NO to indicate that this code is not being used for the
Schedule Access Restriction feature}
CLAS 1 {To allocate code to CLAS 001}
CODE 2465 {Repeat procedure to add code 2465}
SARC
CLAS 1
CODE
REQ END {Exit}
CALL FORWARDING
===============
Introduction
============
On the Meridian system there are a number of features associated with call
forwarding. This section will concentrate on the common ones:
o Call Forward All Calls
o Call Forward No Answer
o Call Forward No Answer, Second Level
o Hunting (on busy)
o Call Forward Busy (Direct Inward Dial (DID) Calls Only)
o Call Forward and Hunt by Call Type
Call Forward All Calls
======================
Description
-----------
This feature is also known as Divert All Calls on other systems.
The Call Foward All Calls feature allows an extension user, to unconditionally
divert their incoming calls to another extension or to an external number.
The forwarding number can be changed at any time by the extension user from
their set. This feature can be assigned to both analogue and digital sets, but if
assigned to a digital set it must be programmed onto a key.
For systems that have Meridian Mail, the extension user can divert their
calls to Mail by keying in the access DN for Meridian Mail.
The feature called "Remote Call Forward" allows an extension user to Program
their Call Forward DN from a remote set.
Implementation
--------------
- Analogue Sets
The Call Forward feature (CFW) is assigned to analogue sets using Overlay
Program 10. The maximum number of digits allowed in the forwarding number
must be specified, the valid entries are 4, 8, 12, 16 or 23.
- Digital Sets
The Call Forward feature (CFW) is assigned to a key on a digital set using
Overlay Program 11. The key number must be specified along with the maximum
number of digits allowed (4, 8, 12, 16 or 23) in the forwarding number.
The extension user can be restricted to forwarding their calls to internal
destinations only by setting a Class Of Service to CFXD (Call Forward
External Denied).
Example
-------
In this example Call Forwarding is assigned to analogue TN 9 3 and Call
Forward External allowed.
>LD 10
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM FTR CFW 16 {Call Forward All Calls to numbers with max. 16
digits}
FTR
ITEM CLS CFXA {Call Forward to external numbers allowed}
ITEM
REQ END {Exit}
On a digital set TN 7 4 Call Forward is to be assigned to key 3 and Call
Forward
External denied.
>LD 11
REQ CHG
TYPE 2008
TN 7 4
ECHG YES
ITEM KEY 3 CFW 8 {Allocate to key 3 Call Forward All Calls to numbers
with
a maximum of 8 digits}
KEY
ITEM CLS CFXD {Call Forward to external numbers denied}
ITEM
REQ END {Exit}
Call Forward No Answer
======================
Description
-----------
This feature is also known as Call Forward On No Reply.
With Call Forward No Answer a call that is not answered after a set number
of rings will automatically forward to an alternative extension.
The feature cannot be activated/deactivated by the extension user and the
forwarding DN cannot be changed by the extension user. The number of rings
before a call diverts is set on a system wide basis between 1 and 15 rings
(4 is the default). For systems that have Meridian Mail, the extension user can
elect to have their calls forward on no answer to Meridian Mail specifying the
access DN for Mail. Call forward no answer will only work single step i.e. once a
call has been forwarded after 4 rings to another extension and it is not answered
at that second extension, it will ring indefinitely unless the extension has
"Call Forward No Answer, Second Level" assigned (see next feature).
Implementation
--------------
The feature is assigned in two steps. First the Class of Service must allow
Call Forward No Answer (FNA), and then the number to which calls are to be
forwarded must be defined. This is known as the Flexible Directory Number
(FDN).
Example
-------
In this example Call Forward No Answer has been assigned to analogue TN 9 3
and the calls will divert to DN 2205.
>LD 10 {Load Program 10}
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM CLS FNA {Allow the Call Forward No Answer feature}
ITEM FTR FDN 2205 {The Flexible DN is 2205}
FTR
ITEM
REQ END {Exit}
In this example Call Forward No Answer has been assigned to digital TN 7 4
and the calls will divert to DN 2205.
>LD 11 {Load Program 11}
REQ CHG
TYPE 2008
TN 7 4
ECHG YES
ITEM CLS FNA {Allow the Call Forward No Answer feature}
ITEM FDN 2205 {The Flexible DN is 2205}
ITEM
REQ END {Exit}
Call Forward No Answer, Second Level
====================================
Description
-----------
Call Forward No Answer will normally only work single step i.e. once a call
has been forwarded on no answer to a second extension it will ring indefinitely
at that extension; if however the second part has a Class of Service Secondary
Call Forwarding Allowed (SFA) then the call will, after a set number of rings,
forward to a third DN as defined in the Flexible Directory Number (FDN) of
the second set.
Implementation
--------------
The feature is assigned to both analogue and digital sets by setting the
Calss of Service to SFA. The FDN must also be defined to identify where the calls
will be forwarded.
Example
-------
In this example Call Forward No Answer, Second Level has been assigned to an
analogue TN 9 3 and the call will be forwarded to 2205.
>LD 10
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM CLS SFA {Allow Second Level Call Forward No Reply}
ITEM
REQ END {Exit}
Hunting
=======
Description
-----------
This feature is also known on other systems as Divert on Busy.
This feature allows a call which encounters a busy extension to
automatically Hunt (forward) to a predefined DN. If the extension that is
hunted to is also busy the system will look at the Hunt DN of that second
set and Hunt (forward) again to that DN.
This continues until an idle DN is found. The Hunt DN is set
on a per extension basis and cannot be changed or disabled by the extension
user.
For systems that have Meridian Mail, the extension user can have their calls
hunt on busy to Meridian Mail in the same way as hunting to another
extension. This is achieved by specifying the access DN for Mail instead of an
extension number. Hunting to Meridian Mail will however preclude the use of some
features like Ring Again.
The maximum number of hunt steps is as follows:
For Option 11, 51, 61 systems = 18 steps
For Option 71 systems = 30 steps
[ On most systems, the hunt number on all the sets will usually be the
extension number of the receptionist or secretary - Snk ]
Implementation
--------------
The feature is assigned to both analogue and digital sets by allowing the
feature in the Class of Service and also defining the Hunt directory number.
Example
-------
In this example Hunting has been assigned to an analogue TN 9 3 with the
calls to be forwarded to DN 2432.
>LD 10
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM HUNT 2432 {Hunt to extension (DN) 2432 when busy}
ITEM CLS HTA {Allow Hunting}
ITEM
REQ END {Exit}
Call Forward Busy
=================
Description
-----------
This feature is set on a per extension basis, and allows for Direct Inward
Dialling (DID) calls only, that encounter a busy extension, to be forwarded
to the Attendant Console if all the steps in the hunting chain are busy.
Implementation
--------------
The feature is assigned to both analogue and digital sets in the Class of
Service.
Example
-------
In this example Call Forward Busy has been assigned to an analogue TN 9 3.
>LD 10
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM CLS FBA {Allow Call Forward Busy}
ITEM
REQ END {Exit}
Call Forward and Hunt by Call Type
==================================
Description
-----------
This feature allows internal and external calls to be routed to different
destination DNs under Call Forward No Answer and Hunting situations.
Implementation
--------------
Note that to implement this feature there are a number of data items,
associated with routes, that cannot be changed by the sysadmin.
Analogue and Digital Sets
-------------------------
The feature is assigned in the Class of Service. Then four directory numbers
must be input to define:
o the DN where internal calls are to forward to on no answer
o the DN where internal calls are to hunt (forward) to on busy
o the DN where external calls are to forward to on no answer
o the DN where external calls are to hunt (forward) to on busy
Example
-------
In this example Call Forward and Hunt by Call Type has been assigned to
analogue
TN 9 3.
>LD 10
REQ CHG
TYPE 500
TN 9 3
ECHG YES
ITEM CLS CFTA {Call Forward and Hunt by Call type allowed}
ITEM FTR FDN 2205 {Destination for Internal calls not answered}
ITEM HUNT 3341 {Destination for Internal calls encountering busy}
ITEM FTR EFD 2731 {Destination for External calls not answered}
ITEM FTR EHT 3421 {Destination for External calls encountering busy}
ITEM
REQ END {Exit}
Call Hold, Permanent
====================
Description
-----------
This feature allows an analogue extension to place an established call on
hold when desired. This feature is not required on digital sets as they have a
dedicated Hold key.
The analogue extension user places the call on hold by following these
steps:
- Press the "Recall" key
- Enter the FFC for Call Hold (#55 is the default)
- Replace the handset
- To take the call off hold and speak to the caller again simply lift the
handset
There is a system wide Permanent Hold Recall Timer which specifies the
maximum length of time that a call can be left in the permanent hold state. If this
timer expires the system will automatically call the user back and reconnect
them to the original caller.
Whilst the caller is on hold they hear silence unless the "Music on Hold" is
equipped. Permanent Hold also requires that the set has the Call Transfer feature
allowed.
Implementation
--------------
The feature must be assigned in Overlay Program 10 and the Class of Service
must allow Call Transfer (XFA).
Example
-------
In this example the Permanent Hold feature has been assigned to TN 9 1.
>LD 10
REQ CHG
TYPE 500
TN 9 1
ECHG YES
ITEM CLS XFA {Transfer allowed}
ITEM FTR PHD {Permanent Hold feature assigned}
ITEM
REQ END {Exit}
Call Party Name Display
=======================
Description
-----------
This feature allows a name of up to 24 characters in length to be associated
with a DN or an access code of a route. When the DN is called it will
display the name associated with the called DN on the calling party's display
module. Analogue sets and digital sets without a display can have a name associated
with their DN that will automatically be sent as they originate calls.
Implemention
------------
The Class of Service for both the calling and called extensions must be set
(in LD 10 or LD 11) to allow Call Party Name Display. The name to be displayed
is set in LD 95, which is also used to print all existing names and DN
associations.
Example
-------
In this example a name has been associated with DN 2302. Note DN 2302 (TN 7
5 ) is assigned to a digital set which has a display module fitted. The example
also shows how a name can be associated with the access code of a route (eg.
1500) to enable extension users to easily identify where calls are coming
from.
>LD 95
REQ NEW
TYPE NAME
CUST 0
DIG {Names can be assigned to "Dial Intercom Group" members,
but just return through this for now}
DN 2302 {DN is specified}
NAME Alan Smith {Name assigned to DN 2302}
XPLN 24 {Set the maximum name length to 24}
DN 1500 {Access code for Route 0}
NAME BT Route 0 {Name assigned to Route 0 (an exchange line)}
XPLN 24
DN {No more names to add so return here}
REQ PRT {Print out names to check}
TYPE NAME
CUST 0
PAGE
DIG
DN ALL {All names are to be printed}
SHRT YES {Short format for printout}
The system responds by printing a list of DNs and names assigned to them:
0 ATTENDANT
1500 BT Route 0
1501 BT Route 1
2202 Janet James
2302 Alan Smith
REQ END {Exit program 95}
Now we need to add the Call Party Name Display feature to the set:
>LD 11
REQ CHG
TYPE 2008
TN 7 5
ECHG YES
ITEM CLS CNDA {Call Party Name Display Allowed. This enables the
displaying of incoming names}
ITEM
REQ END {Exit}
Call Restriction (Barring)
==========================
Description
-----------
- Restriction to Outgoing Routes
Using this process an extension user can be barred from using particular
routes out of the system.
Each outgoing route can be assigned to up to 32 TARGs (Trunk Access
Restriction Groups) which are numbered between 0 and 31. Each extension also has a data
item called TGAR (Trunk Group Access Restriction) to which only one number,
between 0 and 31, may be defined. The extension user is denied from using
the route when the TGAR of that extension matches one of the TARGs of the route.
In the diagram below extension 2731 is barred from using routes 0, 1 and 2.
Extension 2211 is barred from using route 0.
[ ahem, yeah er..sorry about the ascii art (those boxes on the left are the
sets) :] - Snk ]
------ +==============+
| Ex. | +==============+
| 2731 |---------->> | | BT Exchange - Route 0 (TARG 1,4)
------ | P.B.X |---->>-------------------------->>
TGAR 4 | |
| | Private Route 1 to Site B (TARG 2,4)
| |---->>-------------------------->>
| |
------ | | Private Route 2 to Site C (TARG 3,4)
| Ex. |---------->> | |---->>-------------------------->>
| 2211 | +==============+
------ +==============+
TGAR 1
+---------+---------------------------+
| TGAR | Restriction Applied |
+---------+---------------------------+
| TGAR 1 | No access to BT exchange |
| TGAR 2 | No access to site B |
| TGAR 3 | No access to site C |
| TGAR 4 | No access to any routes |
+---------+---------------------------+
- Restriction by Class of Service
Another method of barring calls is by applying one of six alternatives in
the extension's Class of Service.
The six Class of Service types are:
UNR - Unrestricted The extension user can dial anywhere they want -
literally without restriction.
FRE - Fully Restricted No exchange line access. The extension user can
make internal calls, can use privately wired
(Tie)
lines, however they can have calls using exchange
lines transferred to them from another extension
user.
FR1 - Fully Restricted 1 Same as FRE but calls using exchange lines cannot
be transferred to them.
FR2 - Fully Restricted 2 This restricts the extension to internal calls
only.
SRE - Semi-Restricted Allowed outgoing external calls via the attendant
console only. Can recieve external incoming
calls.
TLD - Toll Denied Allowed outgoing calls to certain destinations
only.
The Meridian 1 monitors the digits that are
dialled to determine if the call is to be allowed. This
process is called New Flexible Code Restriction.
- New Flexible Code Restriction
Each individual extension, that has a Class of Service of TLD, will have a
NCOS - Network Class of Service defined. The value assigned to NCOS will
ultimately identify a call barring table (TREE), that contain the numbers that are
permitted or denied from being dialled. Up to 8 trees can be defined on
Meridian 1 and these are identified by NCOS 0 to 7.
Option 11 Default Settings
--------------------------
The default Option 11 system comes with the trees set as follows, however
they can be customised to suite:
NCOS 0 = TREE 0 No Barring (i.e. Unrestricted)
NCOS 1 = TREE 1 Bar Everything except 112 (New European Emergency
Number)
and 999
NCOS 2 = TREE 2 Bar any numbers beginning with 0 or 1 with the
exception
112, 0800 and 0345
NCOS 3 = TREE 3 Bar 010 (International), 0898 and any number beginning
with a 1 with the exception of 112
NCOS 4, 5, & 6 Spare
NCOS 7 = TREE 7 No Barring (i.e. Unrestricted)
Implementation
--------------
The type of call restriction applied to an extension is configured in the
Class of Service using Program 10 (analogue sets) or 11 (digital sets). These
programs are also used to define the extent of teh restriction using the
TGAR and NCOS prompts as appropriate.
Example
-------
This example shows three different extensions being set up for the following
call restriction:
TN 9 1 Totally Unrestricted
TN 8 4 Restricted from gaining access to any route with a TARG of 3
TN 7 4 Restricted from dialling international numbers and 0898 numbers as
detailed in default TREE 3
>LD 10
REQ CHG
TYPE 500
TN 9 1
ECHG YES
ITEM CLS UNR {Class of Service set to Unrestricted}
ITEM TGAR 0 {No Trunk Group Access Restriction is applied}
ITEM
REQ END {Exit}
>LD 11
REQ CHG
TYPE 2616
TN 8 4
ECHG YES
ITEM CLS UNR {Class of Service set to Unrestricted}
ITEM TGAR 3 {Restrict this extension from accessing any route with a
TARG
of 3}
ITEM
REQ
END {Exit}
>LD 11
REQ CHG
TYPE 2008
TN 7 4
ECHG YES
ITEM CLS TLD {Toll Denied causes the system to look at the NCOS value
for this extension}
ITEM NCOS 3 {NCOS 3 causes Code Restriction Tree 3 to be applied to
this
extension}
ITEM TGAR 0 {No trunk group Access Restriction is applied}
ITEM
REQ END {Exit}
Call Transfer
=============
Description
-----------
This feature allows an extension user to transfer an established (answered)
call to another extension. The feature can be assigned to both analogue and
digital extensions.
Implementation
--------------
For analogue sets set the Class of Service to allow Call Transfer (XFA).
For digital sets assign the transfer feature (TRN) to a key.
Example
-------
In this example the Transfer feature has been assigned to analogue TN 9 6.
>LD 10
REQ CHG
TYPE 500
TN 9 6
ECHG YES
ITEM CLS XFA {Allow "Transfer"}
ITEM
REQ END {Exit}
In this example the Transfer feature has been assigned to key 3 of digital
TN 8
2
>LD 11
REQ CHG
TYPE 2008
TN 8 2
ECHG YES
ITEM KEY 3 TRN {Assign "Transfer" feature to key 3}
KEY
ITEM
REQ END {Exit}
Call Waiting
============
Description
-----------
This feature notifies an extension user, currently engaged on an established
call, that another call is waiting to be answered. On a digital set
notification is by a flashing lamp associated with the Call Waiting key and
a buzz tone through the loudspeaker of the set. On analogue sets notification
is by two bursts of tone through the ear-piece of the handset.
The user can put the first call on hold, answer the second call, and return
to the first call when desired. The analogue set uses a FFC, and the digital
extensions must have a dedicated key.
There is an option to notify that internal calls and/or external calls are
waiting.
Implementation
--------------
For analogue sets, allow the Class of Service Call Waiting (CWA), Internal
Call Waiting (SWA) and Warning Tone (WTA).
For digital sets, allow the Class of Service Internal Call Waiting (SWA) and
Warning Tone (WTA). The Call Waiting feature must be allocated to a key.
Example
-------
In this example Call Waiting has been assigned to analogue TN 9 8.
>LD 10
REQ CHG
TYPE 500
TN 9 8
ECHG YES
ITEM CLS CWA SWA WTA {Call Waiting, Internal Call Waiting and
Warning Tone allowed}
ITEM
REQ END {Exit}
In this example Call Waiting has been assigned to key 4 on digital TN 7 0.
>LD 11
REQ CHG
TYPE 2616
TN 7 0
ECHG YES
ITEM CLS SWA WTA {Internal Call Waiting and Warning Tone allowed}
ITEM KEY 4 CWT {Call Waiting allocated to key 4}
KEY
ITEM
REQ END {Exit}
Conference
==========
Description
-----------
This feature allows an extension user to add additional parties to an
established call. The maximum number of parties allowed in conference is
six and only one of the parties may be external.
To establish a conference call the extensiom user would follow these steps:
Analogue Sets Digital Sets
------------------------------------------------------------------------
Press Recall Key Press Conference Key
+ +
Dial the number of the extension that is to be added to the call
+ +
If free, the 3rd party extension rings. After answer a speech link
is established that excludes other conference members. To establish
a conference proceed as follows:
+ +
Press Recall Key again Press the Conference Key again
Repeat these steps to add additional extensions.
Implementation
--------------
For analogue sets the Class of Service must be set to allow Call Transfer
(XFA) and Conference (C6A).
For digital sets Conference (AO6) must be assigned to a key.
Example
-------
In this example the Conference feature has been assigned to analogue TN 9 3.
>LD 10
REQ CHG
TYPE 500
TN 9 6
ECHG YES
ITEM CLS XFA C6A {Transfer and 6 Party Conference allowed}
ITEM
REQ END {Exit}
Distinctive Ringing Groups (DRGs)
=================================
Description
-----------
The Meridian 1 system provides four different ringing candences which may be
assigned to digital sets. The four can best be described in terms of
frequency and rate of change as follows:
DRG Pitch and Rate of Change
---------------------------------------------
1 Low and Fast
2 Low and Slow
3 High and Fast
4 High and Slow
The Meridian 1 system can also br programmed to ring extensions differently
for internal and external calls. This is set up by BT personnel and is set on a
system wide basis.
Implementation
--------------
The Class of Service is set to identify which of the 4 ringing cadences
should be applied to the TN.
Example
-------
This example shows how DRG3 has been assigned to TN 8 3.
>LD 11
REQ CHG
TYPE 2616
TN 8 3
ECHG YES
ITEM CLS DRG3
ITEM
REQ END
Group Call
==========
Description
-----------
This feature allows many, predefined, extensions to be called simultaneously
at the press of a key on a digital set. As each extension answers they
automatically enter the conference call. Analogue sets cannot initiate a
Group Call however they can be included in the group.
The maximum number of members in a group is 20.
The maximum number of groups on a system is 64.
Implementation
--------------
The members in the group (extensions) are defined in a list using Program
18. After the list has been defined the Class of Service, for each extension in
the group must be set to Warning Tone Allowed (WTA) and the feature needs to be
programmed to a key for any digital set that is to initiate the group call.
Example
-------
This example shows how a Group Call List (22) has been set up to call four
extensions (2206, 2401, 2405, 2406). It also shows how the feature has been
assigned to key 4 on TN 7 13.
>LD 18
REQ NEW
TYPE GRP {Define type to Group Call}
CUST 0
GRNO 22 {Define the group number}
GRPC YES {YES indicates that the originator of the group call
controls the call i.e. if the controller clears down,
then
everyone in the group is cleared down. If this field is
set to NO and the originator of the group call clears down
everybody will remain in conference until they clear down
themselves}
STOR 0 2206 {DN 2206 is input into store 0}
STOR 1 2401
STOR 2 2405
STOR 3 2406
STOR {No more extensions to add, so just Return here}
REQ END {Exit}
Now change the Class of Service to WTA for each group member and allocate
the feature to a key for each member that is to originate the group call:
>LD 11
REQ CHG
TYPE 2616
TN 7 13
ECHG YES
ITEM KEY 4 GRC 22 {Assign Group Call for list 22 to key 4}
ITEM CLS WTA {Allow Warning Tone}
ITEM
REQ END {Exit}
EOF.
-------------------------------------------------------------------------------
.
:
|
+--> VoIP <--+
|
:
+--> re-load
VOICE OVER INTERNET PROTOCOL SAID TELEWEST
==========================================
What is VoIP ???
----------------
Voice over IP (VoIP) is one of the most active areas in telecommunications
today. As data traffic is growing much faster than voice traffic, there
has been considerable interest in transporting voice over data networks as
opposed to the more traditional data over voice networks. Most large
carriers are looking at this new technology.
TRADITIONAL TELEPHONY VERSUS VOIP
---------------------------------
The public switched telephone network (PSTN) is the largest network in the
world and is 99.999% reliable. It is a circuit switched system where a
dedicated 64Kb path is set up between the two telephones. This link is
reserved for the duration of the call, whether conversation is taking place
or not, and so is wasteful of bandwidth.
(see included voipnet.jpg)
VoIP is a packet switched system. The phone call is broken into packets of
data, which are sent to the destination using the Internet Protocol (IP).
At the far end the packets are re-assembled and converted back to speech.
Packets from many phone calls, as well as data can travel down the same
path. By using compression techniques and withholding packets containing
silence, further gains in bandwidth utilisation can be achieved.
APPLICATION OF VOIP
-------------------
Voice communications will certainly remain a basic form of interaction
for all of us. The immediate goal for VoIP is to reproduce existing
telephone capabilities but at a significantly lower cost. However,
increased competition in today's telecom market means Telewest is always
looking to provide new services and applications.
VoIP will allow us to introduce innovative new products, quickly and at
competitive rates compared to existing services. Ultimately, wide
deployment of IP telephony will cause a wave of new applications and
services that will fundamentally change the way people use technology to
communicate.
VOIP AND TELEWEST
-----------------
As a first step towards implementing IP telephony, Telewest is currently
conducting a "Proof of Concept Trial" of VoIP over cable within our
Integration Test Platform (ITP) located at Knowsley. Marconi has been
selected to run the trial and supply the backbone infrastructure.
The purpose of this trial is to start Telewest on a learning curve for
this new technology and to assess the ability of our network to support
this new service. If initial test on the ITP prove successful, Telewest
plan to offer the service to a number of trial customers.
Phase two, starting this year, will be to select a preferred solution from
a number of key vendors and roll out a commercial platform in a number of
regional sites for Alpha/Beta trials.
Telewest's ultimate goal is to create a national IP telephony platform
from which we can offer VoIP using all access methods, be it cable, DSL,
or wireless.
Re-LoaD ( #darkcyde idler )
.
:
|
+--> Modem Brown Box <--+
|
:
+--> Phractal
Modem Brown Box
By Phractal.
Using your modem as a bridge for two seperate lines to create '3-way
calling'
This is a common scenario that many dialup users of the interenet have on
their computers. Many people have two phone lines at home, for the
convienience of being connected to the internet via telephone and also being
able to use the other line for voice calls.
In this diagram, we start out with two seperate phone lines, each having
their own unique telephone number of course. One line supplies Phone A with
access to a dialtone. Line two plugs directly into the modem. Almost all
telephone modems have two phone jacks. One is meant for the incoming phone
line (ie the line coming from the wall), and the other is meant to attach to
phone (Phone B), which can be used when the modem is not in use, and will
operate on the same line as line 2, naturally. This is where 'I' and 'O'
labeled on the modem come into place. The 'I' stands for input, and the 'O'
stands for output. On many modems, the input jack is labeled 'Line', and the
output jack is labeled 'Phone'. When the modem is not in use, think of it as
a router for Line 2 to Phone B.
*POTS lines from wall
Line 1 Line 2
Phone Line Modem Line
|~| |~|
| |
| |
| | /----------\
| | | |
| | /-------------\ | Computer | /----------\
| | | | | | | @@@@@@@@ |
_______ \-------|<I | | | | @@@@@@@@ |
######### | Modem |----------| | | @@@@@@@@ |
###|___|### /-------|<O | | | \__________/
* 123 * | | | | |---\ /
* 456 * | \-------------/ | | _/______\_
* 789 * | | |
*********** | \----------/
Phone A |
|
_______
#########
###|___|###
* 123 *
* 456 *
* 789 *
***********
Phone B
With some tinkering, all the materials described and illustrated above can
be
used to create a bridge of two seperate phone lines. This means that you can
call two seperate phone numbers and communicate with the people on the
recieving line at the same time, and they can hear you, and they can hear
each other, and not just you. In laymen's terms, it's three way calling. It
isn't really three way calling, because actual three calling is a service
offered by the telephone company to a single phone line, but this plan has
the exact same effect as actual three way calling. It is cheaper to have one
phone line with the three way calling plan, than to have two phone lines,
but
if you have/need/prefer two phone lines, you can use this method to make
three way calling using two phone lines. This is totally legal as well. You
can do this same procdedure with a phone that accepts 2 lines.
To achieve this method of three way calling, setup your modem, phones and
phone lines in the following manner:
*POTS lines from wall
Line 1 Line 2
Phone Line Modem Line
|~| |~|
| | /Note: Line 1 is connected to wall, and
| | / unplugged at other end
| | / /----------\
| | / | |
| | / /-------------\ | Computer | /----------\
| | | | | | | | @@@@@@@@ |
\-----------|----- |<I | | | | @@@@@@@@ |
| | Modem |---------| | | @@@@@@@@ |
\--------|<O | | | \__________/
| | | |---\ /
\-------------/ | | _/______\_
| |
\----------/
Phone A Phone B
_______ _______
######### #########
###|___|### ###|___|###
* 123 * * 123 *
* 456 * * 456 *
* 789 * * 789 *
*********** ***********
| |
| |
| |
| |
| |
|~| |~|
Line 1 Line 2
Phone A and B are both unplugged from their previous destinations. They need
to be plugged into other telephone jack terminals for Line 1 and Line 2. Be
in mind, these are still the same Lines that were mentioned before. At the
original terminal, Line 1 directly from the wall is now unplugged (but get
ready to plug into Input/Line modem jack on modem), and directly from the
wall, line two is plugged into the the Phone or Output jack.
Instructions for three way calling operation:
1) Pick up Phone B connected to Line 2 and hear a dialtone. Dial a phone
number of someone you want to speak to.
2) While talking on Phone B once they've picked up and you've confirmed a
connection, plug line 1 into the Input/Line jack of the modem.
3) You should hear a dialtone, but you are still also connected to your
friend who you previously called. You can still hear eachother if you speak
loud enough over the dialtone. This dialtone is the dialtone of Line 1.
4) Pick up Phone A, which is connected to Line 1 and dial a number of
someone
you wish to speak to. You should hear the tones of the numbers being dialed
on Line 2/Phone B, as well as Phone A of course. The dialtone will stop
as it always does once you enter a DTMF tone.
5) If all goes well, and the third party picks up, both of the people can
talk to you, hear you, and hear eachother, hence three way calling.
***NOTE***
With step 4, you can dial the third party directly from your second line,
or
in this case, Phone B. I told you to dial from Phone A because it better
illustrates how this works.
Also, reset to original setup to get online :)
~~~Why this works.~~~
The Output jack for the modem also accepts input over a phone line, because
otherwise, how would you talk to someone on a phone routed through a modem,
without them hearing you. Your voice is the input.
The Modem thinks that the lines you connected in diagram 2 are one phone
line. And since the Line 2 is connected to the output modem jack and then to
the wall, instead of an individual phone, the modem is now in essence,
'connected' with the rest of the phones all hooked up to line 2.
The thing that triggers the whole thing is when you connect line 1 to the
modem. Now, I'm not totally sure why this part happens, but I think it's the
modem trying to figure all of this out. So here's my hypothesis: The modem
recognizes that there is activity on Line 2, so it opens up the Input jack
for information automatically, and therefore, you get an auto-dialtone when
you insert Line 1 into the 'Line/Input' Modem jack. The modem is basically
taking the phone off the hook on the Input jack. And it's all ready for you
once you insert Line 1.
Slackware ownz!
--Phractal
.
:
|
+--> TDM <--+
|
:
+--> foneman
--------------------------------------------------------------------------
Time-Division Multiplexing
by foneman
--------------------------------------------------------------------------
The problem of transmitting multiple signals over a single-wire
circuit was one of the earliest challenges to telecommunications
inventors. It was pretty clear that this challenge needed to be conquered
seeing as how costly it was to install wire circuits. The solution was
sought after to solve the problem for sending multiple telegraph signals
over a single-wire circuit.
One early way to transmit more than one signal over a single-wire
circuit was by use of a phantom circuit. Over already existing wire pair
circuits, an additional signal was transmitted through the use of
transformers at each end of the circuit. This type of signal was known as
a phantom signal.
Although the phantom circuit solved the problem of transmitting
more than one signal over a wire circuit, it only solved it to some
extent. The only way more phantom signals could be transmitted over the
wire circuit pair was if the technique was extended to more wire
circuits, which was still very costly and still only allowed one phantom
signal per pair circuits.
Multiplexers
------------
Eventually multiplexing was invented. A multiplexer, often called
a mux, allows several devices to transmit a signal over the same
transmission medium. For example, if you have a high-speed circuit and
you wanted to run several low-speed terminals off of it you could run the
terminals through a multiplexer. This is obviously a major advantage.
Multiplexers are normally used in pairs, with one at each end of
the communications circuit. The multiplexer connected to the terminals
can combine the data and send it over the communications circuit, then the
multiplexer on the receiving end would the separate the data and send it
to the correct ports.
Before I get into multiplexing, a few things need to be said so
that the rest of the article is understood. From this point on a
multiplexer will be refered to as a mux, as it often is. The low-speed
ports of a mux will be refered to as branch ports and the high-speed port
as the trunk port. The branch ports of a mux are usually connected to
terminals or front end ports and the trunk port to the communications
circuit. Two muxes connected by a communications circuit is illustrated
below to give you a visual if you had any trouble with that.
Mux A Mux B
___________ ___________
| | | |
* | | *
* #=========================# *
* | | *
* | | *
|___________| |___________|
= Communications circuit
* Branch port
# Trunk Port
Note: The two muxes shown in the above diagram are only connected to
each other. In a realistic situation the branch ports would
probably be connected to terminals on the sending end and a front
end processor, or FEP, on the receiving end.
Pure Time-Division Multiplexing
-------------------------------
When you have a high-speed communications circuit that's being
used by several low-speed devices the muxes use a technique known as time-
division multiplexing, or TDM. In my first example of TDM, shown in the
diagram below, four terminals and front end ports are operating at 1200
bps. The modems between the muxes each operate at 4800 bps. All four
terminals and front end ports share the same bandwidth of 4800 bps.
Host computer
_____________ Terminals
| | ___
| | FEP _| |
| | ________ Mux Mux | |___|
| | | | ______ Modem Modem ______ | ___
| | | |--| | ____ ____ | |-- | |
| | | |--| |_|____|_____|____|_| |----|___|
| |___| |--| | | |-----___
|_____________| |________|--|______| |______|-- | |
| |___|
_|_
| |
|___|
Obviously you can see that if all four terminals were constantly
transmitting and receiving data, there would be a total of 4800 bps
transmitted and received by the modems. Since there is a 4800 bps
connection between the modems and only four terminals and front end ports
operating at 1200 bps there will always be enough bandwidth to handle
all of the terminals' maximum operations.
How does Pure Time-Division Multiplexing Work?
----------------------------------------------
At this point I'll discuss the two most popular methods of pure
time-division multiplexing; character interleaving and bit interleaving.
In character interleaving, also known as byte interleaving, one
character is multiplexed at a time. A diagram will be given below to help
you to better understand what's being explained in this section. So
things don't get messy, only four terminals (each sending a one word
message simulatiously) will be used. Terminal 1 sends "Phreaker,"
terminal 2 sends "Linux," terminal 3 sends "Fetus," and terminal 4 sends
"Ajax"; all words are sent one character at a time by the terminals. The
mux connected to the terminals will be mux A, and the mux connected to the
front end processor will be mux B.
Using character interleaving, mux A scans it's branch ports for
characters transmitted by the terminals. Should mux A receive any
characters to be sent over the high-speed trunk port, they can be sent
over it one at a time. For example, mux A scans it's branch ports and
discovers the characters P, L, F, and A. Mux A sends these characters
over it's trunk port to mux B. Mux B then sends the characters to each
front end port at 2400 bps. The P is sent to port 1, the L to port 2, the
F to port 3, and the A to port 4. Next, mux A sends 'hiej' over its trunk
port. Mux B then sends h to port 1, i to port 2, e to port 3, and j to
port 4. This goes on until every character is sent.
Host computer
_____________ Terminals
| | ___
| | FEP _| 1 |
| | ________ Mux B Mux A | |___|
| | | | ______ Modem Modem ______ | ___
| | | |--| | ____ ___ _ | |-- | 2 |
| | | |--| |_|____|____|____|_*| |----|___|
| |___| |--| | | |-----___
|_____________| |________|--|______| |______|-- | 3 |
| |___|
_|_
| 4 |
|___|
* Byte interleaving time slots and their data
Time
slot: 12341234123412341234123412341234
Data: PLFAhiejrntaeuuxaxs-k---e---r---
Every time a front end port is communicating it's called a time
slot. In the example I just gave, terminal 1 had the first time slot,
terminal 2 had the second time slot, terminal 3 the third, and terminal
4 the fourth. When the data reaches mux B, the first time slot is sent
to front end port 1, the character in the second time slot is sent to
front end port 2, and so on. However many devices are used with TDM,
there would be that many time slots instead of four. For example, if
there were 10 terminals instead of just four, there would be 10 time
slots.
Time slots aren't always used, as the example of terminal 4
sending the message "Ajax" shows. This just means that bandwidth goes
unused. Every terminal is guaranteed enough bandwidth because the speed
of the trunk port equals that of the speeds of the terminals combined.
The other mentioned method of accomplishing TDM is bit
interleaving. As opposed to sending one character from each terminal,
this method takes one bit from each terminal and sends it on the trunk
port. Imagine that terminal 1 sends the character F (01000110), terminal
2 sends O (01001111), terminal 3 sends N (01001110), and terminal 4 sends
a E (01000101). We'll ignore start and stop bits to make it simpler.
Since in ASCII the bit on the right is transmitted first, the first bit
received from terminal 1 is 0, from terminal 2 is a 1, from terminal 3 is
a 0, and from terminal 4 is a 1. 0101 is then sent by mux A over the
trunk port. Mux B receives the data and sends the first time slot to
front end port 1, the second time slot to front end port 2, the third time
slot to front end port 3, and the fourth time slot to front end port 4.
Since in bit interleaving, the time slots are shorter, there are eight
times as many time slots in a second. In this case too, the speed of the
trunk port has to be equal to the speeds of the terminals combined. The
two methods function identically, except for the fact that there is a bit
in each slot instead of a byte.
* Bit interleaving time slots and their data
Time
slot: 12341234123412341234123412341234
Data: 01011110111101100000000011110000
Statistical Time-Division Multiplexing
--------------------------------------
Due to the fact that humans take breaks, screw around on the job,
and type slower than their terminals can transmit, users generally don't
use all of the available bandwidth. Because of those reasons, character
transmission is usually very intermittent. In a situation where this
happens more often than not using pure TDM, time slots are left empty.
Statistical time-division multiplexing (STDM) takes full advantage
of the intermittent nature of terminal users and does it's best to make
available bandwidth to each terminal when needed. When using an STDM mux,
also called a stat mux, the combined speeds of the terminals in use do not
have to equal that of the transmission speed in the trunk port. For
example, six terminals at speeds of 2400 bps can use a trunk port
transmitting at 9600 bps. This is due to the nature of STDM in that it
assumes not all devices are transmitting all the time.
Since in STDM, time slots are not sent on a constant basis the
question of how the stat mux knows which data goes to which port?
Typically, a stat mux sends an address along with the data. When a
terminal is not sending any data, other terminals transmitting can use the
time slots.
Another question comes to mind when speaking of STDM. What
happens when all terminal users attempt to transmit data at the same time?
Since the data can't all be transmitted at once, a stat mux will generally
buffer the data until a terminal user stops sending data so that there
would be extra time slots available on the trunk.
Closing
-------
Obviously you can see which method of TDM would be most efficient
in a case of intermittent transmission or in constant transmission. I
hope that I was able to shed a little light on what's happening to your
data as it's sent.
foneman
.
:
|
+--> UK Trunking Network <--+
|
:
+--> hybrid
UK Trunking Network Primer.
by hybrid <hybrid@f41th.com>
Written for F41th and 9x April 2001
-----------------------------------
The Local Distribution Network (Local Loop)
===========================================
At the lowest level in the UK PSTN hierarchy, customers Terminal Equipment
(TE) is connected to the Local Loop, which physicaly comprises of cables
and copper. The Local Distribution Network provides the TE with access to
the Local Exchange either via a standrard copper connection, where the
connection to the exchange is analogue, and then conversion from the
exchange to the PSTN is digital, or via a direct digital link (FAS
Flexible Access Systems) from the TE to the Local Exchange.
A Logical Local Distribution Network
====================================
P <--> +----------+ +-------+ Exch Side +-------+
S <--> | Local | +------>| PCP |<---------->| SCP |
T <--> | Exchange | | ###| |############+-o-o-o-+
N <--> | MDF | | # +-o-o-o-+ Dist Side | | |
+-------+--+ | # | | | | |
| CC | | # | | | |
| |<------+ # | | |
+-------+ # | | +---------+
############################## | +---------+ |
+--------------+ | |
| | |
| | |
+------R+D------+ +------F+T-+ +-O+D------+
| | | | | | |
| | | | | | |
+--+-+ +--+-+ +--+-+ +--+-+ +--+-+ +--+-+ +--+-+
| TE | | TE | | TE | | TE | | TE | | TE | | TE |
+----+ +----+ +----+ +----+ +----+ +----+ +----+
RD = Radial Distribution (Underground Cables)
FT = Frontage Tee Distribution
OD = Overhead Distribution
CC = Cable Chamber
PCP = Primary Cross - Connect Point
SCP = Secondary Cross - Connect Point
MDF = Main Distribution Frame
### = Exchange Side/Distribution Side Barrier
UK PSTN Node Hierarchy
======================
The UK PSTN is a complex array of network nodes that operate on a
partent-child hirarchy, operating right from the top of the chain
(International Exchange Boundarys) down to RCUs and RSSs to customers TE.
BT implement a particular system known as System X, which will be covered
later in this file. A basic layout of the UK PSN is as follows:
+---------------+
| International |
| Gateway |
+---+---+-------+ DDSSC = Digital Derived
| | Services Switching
| +-------------------+ Centre
| |
+---+---+ +---+---+ +---------+ --> D
| DMSU O<------------->O DMSU O<-------->O| DDSSC | --> D
| | | | | | --> S
+-+---+-+ +-+---+-+ +---------+ --> N
| | | |
| | +---------+ | DMSU = Digital Main
| | | | Swithing Unit
| | | | (System X)
| +-------------------+ | DLE = Digital Local
| | | | Exchange (System X
| | | | or AXE10)
| +---------+ | | RCU = Remote Concentrator
| | | | Unit (System X)
+-+---+-+ +-+---+-+ RSS = Remote Subscriber
| DLE | | DLE | Switch (AXE10)
| O<--------+ +-->O |
+---+-+-+ | | +---+-+-+
| | | | | |
| | | | | +-------------+
| | | | +------------+ |
| | +-o-o-----------+ | |
| | | Operator | | |
| | | Service Centre| | |
| | +---------------+ | |
| +---------+ +------------+ |
| | | +--+
+---+---+ | +---+---+ |
| RSS | | | RCU | |
| | | | | |
+---+---+ | +---+---+ |
| | | |
| | | |
\ TEs / \ TEs /
The International Gateway
=========================
The International Gateway, or ISC (International Switching Centre) is a
switching centre that will handle international traffic to and from other
countries. In the UK There are 5 ISCs that implement AXE10, 5ESS and
DMS100 switches. As shown in the diagram above, the ISCs are connected to
the inland PSN via System X DMSUs.
The UK Trunk Network
====================
ISCs are connected to DMSUs (Digital Main Switching Units) which form the
backbone of the UK trunking network. There are approximatly 59+ DMSUs in
the UK PSTN which are all fully interconnected and switch/route traffic to
and from System X catchment areas, along with around 4 Digital Switching
Units (DSUs) which route traffic from London to ISCs via partially
interconnected links.
The DMSUs are interconnected to the "smaller" nodes in the Local network,
forming the connection from the national switched PSN to the Local Level.
Aswell as being interconnected to the ISCs, the DMSUs are also connected
with the DDSN (Digital Derived Services Network) which provides BT
LinkLine services, such as 0800s.
The UK Local Network
====================
At the top part of the Local Network are the DLSUs (Digital Local
Switching Units) which from the local level in the System X network.
A zone will normaly contain more than 2 DLSUs (System X) and/or AXE10
Exchanges. From the 2+ Exchanges in the zone, 1 of them is used as a DCCE
(Digital Cell Centre Exchange) or a DLE (Digital Local Exchange).
So a DLSU will either be a DLE or a DCCE coupled with a non System X
(AXE10 Exchange), Otherwise known as System Y.
The DCCE has access to its parent (the DMSU) via its DDF (Digital
Distribution Frame). Any other Exchange on this network level, such as
AXE10 must also implement the DCCEs Digital Distribution Frame as a means
of access to and from the parent DMSU.
The DCCE performs variopus functions at a local level, including call
charging and Tandem Traffic Switching between other DLSUs on the local
network level. The DLSU acts as a parent node for the next type of local
exchange, the RCUs (Remote Concentrator Units).
In cases where older ALEs (Analouge Local Exchanges) have been replaced
with RCUs a DLE will replace the DCCE and and act as a parent for the
collection of replaced ALEs forming an RCC (Remote Concentrator Centre).
When there are more than one Concentrators in an RCC, this will form an
RMCC (Remote Multi Concentrator Unit)
Because older ALEs such as TXE4 have no direct interconnection with the
main DMSUs, the DCCE will provide them with the function to communicate
with the DMSU. It is possible to determine your own Local Exchange type
from your TE (Terminal Equipment(Phone)) via Testing the Exchange for
Offered Services:
System X: A System X type is the only Digital Exchange to offer "Charge
Advise Per Call" *40*[DN]#. Also "Regular Reminder Call" *56*[hh/mm]*9# so
if your Excahnge excepts these 2 services, it is most likely System X.
System Y: "Charge Advice Per Call" this time, *40#[DN] will determine the
System Y Exchange Type. If neither of these work, there is a
possiblilty that your exchange type is UXD5(Non C7 or Enhanced).
In some high call density zones in London, a DJSU (Digital Junction
Switching Unit) to handle the high concentration. The TE Customers are
connected to a RCU/RSS/ALE via the Local Loop, and the RCU is connected to
the DLSU via digital links. The opposite of this is Low density rural
areas whereas UXD5 is used.
######TRUNKING#LEVEL#################################
... <------>ISCs<------><----------->DDSSCs-----> ...
... <------>DMSU<------->DMSU<------>DMSU<------> ...
| | |
| | |
######LOCAL#LEVEL####################################
| | |
DLSU DLSU DLSU<---> Other DMSU
DCCE DLE DCCE
| | |
RCU RCU RCU (or RSS AXE10)
ALE RCU
ALE
PCM
===
Pulse Code Modulation forms the backbone of the UK digital network,
including direct links to and from digital Exchanges. One of the main
functions of PCM is to convert reproduced analouge voice signals into
a digitaly pulsed form, which can then be transmitted over the UK digital
Trunking Network.
The first part of the PCM proccess is to sample the origional transmission
into PAM (Pulse Amplitude Modulation) Signals. This is done by sampling
the voice pattern at fast intervals (8kHz on the UK network) into
real-time PAM signals which are then encoded into the PCM signal. In order
for the PAM pulses to be encoded, they must be treated as strings/samples,
and then injected with a standard 8 bit binary string.
When the 8 bit binary strings arive at the other side, ie: the destination
of the decoder on the network, the binary samples can be decoded back into
origional form, where the reconstruction of the PAM signals takes place.
Now, because of the time intervals in the sampling rate of conversion,
this will leave Spaces for other signals to be injected into the
transmission, thus multiplexing occurs. This is more commonly known as TDM
(Time Division Multiplexing), where each sample will take a Timeslot. In
PCM the samples are converted at a rate of 8kHz, thus creating 32
Timeslots.
BT implements a 30 CHANNEL Timeslot system, in which the remnaining 2
CHANNELS are used for Network Syncronisation (TIMESLOT 0) and Network
Signalling (TIMESLOT 16), ((Loop Disconnect (LD) type signalling)).
During each frame two sets of four signalling bits, (a,b,c,d) are
allocated to carry signalling information for 2 traffic channels.
Frame Signalling Codes:
=======================
a b c d
--------------------------------------------------------------------
1 1 1 1 Circuit Idle Circuit Busy
0 0 1 1 Circuit Seized Answer
1 0 1 1 Dial Break Not Used
0 1 1 1 Not Used Circuit Free
0 0 0 1 Operator Override Not Used
(The Operator Override code is a
forward signal used to allow an
operator to monitor a busy line)
1 0 0 1 Not Used Not Used
1 1 0 1 Not Used Not Used
0 1 0 1 Not Used Not Used
0 0 0 0 Not Used Not Used
--------------------------------------------------------------------
+-------+
|TS 0 |< Network
| |< Synchronisation
+-------+<-----+
|TS 1 | |
+-------+ |
|TS 2 | |
+-------+ |
|TS 3 | +- Speech Channels
| . | | 1 to 15
| . | |
|TS 14 | |
+-------+ |
|TS 15 | |
+-------+<-----+
|TS 16 |< Network
| |< Signalling + Allignment Signals
+-------+<-----+
|TS 17 | |
+-------+ |
|TS 18 | |
| . | |
| . | +- Speech Channels
|TS 29 | | 16 to 30
+-------+ |
|TS 30 | | CAS (Channel Associated Signalling)
+-------+ | ===================================
|TS 31 | |
+-------+<-----+
Timeslot 0 is injected with allignment signals in the transmission path of
the PCM signal, the PCM decoder will then be able to know if the
synchronisation rate is correct. The following is a diagram of a PCM
encoder/decoder circuit which is implemented in this proccess:
PCM CIRCUIT
===========
+<-- Ch (4)
+---+ |
| |-------------->+|<-- Ch (3)
+---+ ||
Low Pass Filters ||
+---+ ||
| |------------->+||<-- Ch (2) Receive End --------->
+---+ ||| +--> Ch (16)
Ch (1) ||| |
+---+ || ||| +-----------------------+ +|--> No Channel
#-#-->| |-||--->+||+->O | || Connected
AUDIO +---+ || ||+-->O | +||--> Ch (15)
|+--->O | |||
------------------+---->O<----------+<-TS (0) O<----+||
Reconstructed +------->O | TS (16)->O<-----+|
PAM Samples |+------>O | O<------+
|| +-----------|-----------+
---------------+| Ch (1)----->| Recognised Signals
No Channel | Ch (3)----->|<---- Ch (2)
Connected | |<---- Ch (4)
| |<---------------------+
----------------+ +-----------|-----------+ | C
Ch (30) | | | o
| DECODER | | m
| | | m
+-------------|---------+ | o
. .+........................... . +---+ B P | n
. Ch | 1 | i A |
. +---+ n M | P
. Ch | 2 | a | a
Line +---+ r S | t
System Ch | 3 | y i | h
. +---+ g |
. Ch | 4 | n <----------+ F
. +---+ a | o
. | | l | r
. .+........................... . +---+ s |
+-------------|---------+ | A
| | | l
| ENCODER | | l
| | |
+-----------|-----------+ | C
Ch (1)----->| PAM Samples | h
Ch (3)----->|<---- Ch (2) | a
+<-- No Channel |<---- Ch (4) | n
| Connected |<---------------------+ n
| +-----------|-----------+ e
| Ch (30) ---------->O | | l
+---------------------->O | | s
+--------------->O<----------+<-- TS (0) |
| +--->O O<------+
| |+-->O Channel Sampling O<-----+|<-- TS (16)
=A====== ||+->O Switch O<----+||
==U===== ||| +-----------------------+ |||
===D==== ||| |||
====I=== ||| Transmit |||
=====O== ||| End |||
======|= ||+-> Ch (4) |||
| |+--> Ch (3) Ch (16) <----||+
| +---> Ch (2) No Ch <----|+
+---------------> Ch (1) Ch (15) <----+
Timeslot 16 and CCS (Common Channel Signalling)
===============================================
As with CAS, in CCS Timeslot 16 is used for network signalling information
such as binary sieze codes etc. In CCS, Timeslot 16 is a 64/kbit channel,
(8 bits x 8 kHz), as are the other channels.
Digital Exchanges implement CCS for signalling between various other
Exchanges, and is based on CCITT7. In System X, the 30 Channel PCM system
forms the link between the customer and the Exchange switch (DS3), and
from the Excahnge switch to Junction Access. At the Concentrator Unit in
the RCC, subscribers lines are concentrated into PCM Channels, upto 2048
lines consentrated to the 8 x 30 PCM channels in each unit. DSSS
Subscriber Line Units at the Concentrator:
Sub 0 +-------+ Speech +-------+
------->O O<------------->O L C |
| SLU 0 | | i o |
------->O O<------------->O n n |
Sun 7 +-------+ Signals | e t | +---------------+
| r | 32TS's| Concentrator |
| o O<----->O Switch |
| l | | |
| l | | |
Sub 24 +-------+ Speech | e | +---------------+
------->O O<------------->O r O<--+
| SLU 3 | | | |
------->O O<------------->O | |
Sub 31 +-------+ Signals +-------+ +--> Concentrator
Control
If the Remote Concentrator becomes Isolated from the Exchange, ie: unable
to communicate with the Exchange, Subscribers are still able to set up
calls, although calls made in this period of isolation will not be charged
due to the charging equipment in the Exchange dependancy on the
concentrator.
Review
======
IAM Initial Address Message
IFAM Initial and Final Address Message
ISC International Switching Centre
|--> Handle traffic to and from other countries
|--> 5 ISCs in the UK:
|--> Keybridge
| |--> AXE10
|--> Kelvin
| |--> AXE10
|--> Mondial
| |--> 5ESS
|--> Madley(A)
| |--> AXE10
|--> Madley(B)
| |--> DMS100
|--> Connected to Inland PSTN via DMSUs
ISDN Intergrated Services Digital Network
|--> ISDN lines tested from dedicated OMC line test system
ISUP ISDN User Part
IUP Interconnect User Part
MTP Message Transfer Part
PNO Public Network Operator
PSTN Public Switched Telephony Network
SCCP Signalling Connection Control Part
TUP Telephony User Part
ETSI European Technical Standards Institute
NCRS Non Circuit Related Signalling
GSM Global Systems for Mobile
DSU Digital Switching Unit
|--> DSUs are found in London
|--> Trunk and Network traffic switched to London Boundrys
|--> DSUs are only connected to DMSUs in London
DMSU Digital Main Switching Unit
|--> Parent of DLSU
|--> Fully Integrated Trunk Network
|--> Switch Traffic to and from Catchment Area
|--> System X
|--> Manufactured by GEC/Plessey (GPT)
DLSU Digital Local Switching Unit
|--> Digital Local Processor Exchange
|--> Either a DCCE (+) DLE
|--> System X (+) AXE10
DCCE Digital Cell Centre Exchange
|--> Acts as parent for RCUs
|--> Acts as parent for Analouge Exchanges
| |--> Terminates Analouge Line plants
| |--> Executes call charging
|--> Switches Tandem traffic between DLSUs
|--> Access to DMSU via DDF (Digital Distribution Frame)
DJSU Digital Junction Switching Unit
|--> Used in High Density Telecom Areas
|--> Tandem Switching Functions
|--> No Direct Customer Connections
|--> Only in London
|--> System X Exchange Type
DSSS Digital Subscribers Switching Subsystem
|--> Refered to as "DS3" or "D triple S"
|--> Consists of Several Concentrators
|--> P-Switch (Part of DSSS)
DDSN Digital Derived Services Network
|--> Connected to the PSTN
|--> Provides LinkLine Services 0800/0345 etc
|--> Controled by DDSSCs
| (Digital Derived Services Switching Centre)
| |--> which in turn are controled by:
| |--> INDB (Intelligent Network Database)
|--> Connects Service Providers to PSTN via DMSUs
ALSU Analogue Local Switching Unit
|--> Analogue Switched Functions to Customers
| |--> Strowger (TXS)
| |--> Crossbar (TXK1)
| |--> Electronic (TXE2, TXE4)
|--> TXE4 system enhanced to implement digital
|--> CCS (Common Channel Signalling)
ALE Analouge Local Exchange
|--> No direct DMSU Links
|--> Traffic Forwarded to DMSUs via DCCE
DLE Digital Local Exchange
|--> A DLSU that hosts RCU on a hierarchial relationship
|--> Parent of ALEs replaced by RCUs
RCU Remote Concentrator Unit
|--> Part of the DLSU via connection
|--> System X Concentrator
| |--> Concentates large number of lines
| |--> Lesser number of PCM channels
|--> Maximum number of Terminated lines: 2048 (0-2047)
|--> MUST be connected to DLSU
RSS Remote Subscriber Switch
|--> Same as RCU, but AXE10
RCC Remote Concentrator Centre
|--> Multiple RCUs (+) RSSs at one location
|--> (RMCC) Remote Multi Concentrator Unit
UXD5 CDSS1 (Monarch PABX) Derived Digital Exchange
|--> Some depraved Rural areas (being phased out)
5ESS Used in DDSN Network and International Gateway Exchanges
DMS Used in International Gateway Exchanges and BT Featurenet
AXE10 System Y
|--> IOG (Input Output Group)
| |--> SPS (Support Processor Subsystem)
| |--> MCS (Man-Machine Communication Subsystems
| | |--> Alarm Panels, Terminals etc
| |--> DCS (Data Communications Subsystem)
| | |--> Communications over digital links
| | | |--> OMC
| | |--> Same as AUS/NIS in System X
| |--> FMS (File Management Subsystem)
|--> APT
| |--> SSS (Subscriber Switching Subsystem)
| | |--> Same Concentrator function as DSSS
| | |--> LSS Local Subscriber Switch
| | |--> RSS Remote Subscriber Switch
| |--> TSS (Trunk and Signalling Subsystem)
| | |--> Handles connections to other nodes
| |--> GSS (Group Switching Subsystem)
| | |--> Connects speech paths like DSS
| |--> CCS (Common Channel Subsystem CCITT7)
|--> APZ
|--> RPS (Regional Processing Subsystem)
| |--> Scan subscriber lines
| |--> Operation of Switches
|--> CPS (Central Processing Subsystem)
|--> 2 Central Processors (like PUS)
TE Terminating Equipment
|--> Terminate Customer Line, isolate Exchange
|--> Terminating point for each junction
OMC Operations and Maintenance Centre
|--> Serves Partitioned Digital Exhanges
|--> Data collected from Exchages and served to users
|--> OMU
|--> Traffic
OMU Operations and Maintenance Unit
|--> Office with engineers monitoring Exchanges
| |--> Recieving data from the OMC
|--> Multiple OMUs connected to 1 OMC
|--> Connected to UXD5 Exchanges via stand-alone terminal
|--> Terminals directly connected to the OMC
| |--> Control over Exchanges connected to the OMC
|--> Bottom of hierarchy
|--> System Manager
|--> UM (Operations and Maintenance Unit Manager)
|--> MCO (Maintenance Control Officer)
| |--> TCD (Task Co-ordinating Duty)
|--> MO (Maintenance Officer)
|--> MOA (Maintenance Officer A)
|--> MOB (Maintenance Officer B)
DSG District Support Group
|--> Bellow PSG
|--> BT District Staff
PSG Product Support Group
|--> Bellow ISOS
|--> BT Headquaters
ISOS In Service Operational Support
|--> Top of hierarchy
|--> Manufactures
EIR Exchange Incident Report database
SLU Subscriber Line Unit
TNS Test Network Subsystem
LCP Local Control Point
GPA General Purpose Test Auxiliaries
|--> Monitor
|--> Looping and Ring Trip
P-Switch Part of DSSS. Analouge switch for concentrating
physical test access from the subscriber line to
the:
Q-Switch Second Switching Stage for subscriber access
|--> built from 8x8 2 wire reed switch SIUs
R-Switch Access to short handeling time access auxilaries
|--> built from 3x3 2 wire reed switch SIUs
D-Switch Connects the user into the Test Network
|--> Access gained via dedicated DIAL BACK circuit
MTN Mini Test Network
|--> Test subscriber lines and equipment
TAC Test Access Connection
PLA Per Line Auxiliarys
MDF Main Distribution Frame
TOS Temporarily Out Of Service
|--> Will still give dialtone when siezed
| |--> only 999 and 151 calls can be made
|--> To confirm line has been TOS'd make test call to
TOS'd line and listen to message, or from the TOS'd
line call an un-charged number and await NU tone
FT Failure Tone
|--> 600Hz at 260Ms on, 260Ms off, 1400Hz at 260Ms off)
CPI Cable Pair Identification
PCP Primary Cross connect Point
|--> SCP (Secondary Cross connect Point)
FAS Flexible Access System
TRS Transmission Repeater Station
PSN Public Switched Network
OSC Operator Services Centre
VPN Virtual Private Network
|--> BT Featurenet
PC Private Circuit Network
ACE Automatic Cross-connection Equipment
PDN Public Data Network
|--> PSS (Packet Switch Stream)
OLO Other Licenced Operator
VAN Value Added Network
|--> BT Cellnet
PCM Pulse Code Modulation
|--> CAS (Channel Associated Signalling)
|--> CCS (Common Channel Signalling)
PAM Pulse Amplitude Modulation
TDM Time Division Multiplexing
MML Man Machine Language
PUS Processor Utility Subsystem (System X main Processor)
DSS Digital Switching Subsystem (Main System X Switch)
MTS Message Transmission System (C7 Messages)
SIS Signalling Internetworking Subsystem (Analouge)
AUS Access Utility Subsystem (System X)
NIS Network Interworking Subsystem (OMC Access and Admin)
COU Central Operations Unit
|--> National Operational functions are performed here
|--> Network Traffic Managment
NOU Network Operations Unit
|--> UK Network is divided into 9 zones
| |--> Scotland
| |--> North East
| |--> North West
| |--> N.Ireland
| |--> Midlands
| |--> Wales and West
| |--> Southern Home Counties
| |--> Northern Home Counties
| |--> London
|--> Remote Manipulation of the Network
|--> Circuit Provisioning
|--> Control of work between NOU and NFU boundarys
NFU Network Field Unit
|--> Physical Network access (Engineers etc)
NACC Network Administration Computer Center
|--> 11 in NOU Zones around the UK
|--> 1 in Martlesham for emergency fallback
|--> NOMS (Network Operations Managment Systems)
|--> CSS (Customer Services System)
CTU BT Circuit Termination Unit
|--> located within the building housing the Operator
|--> G703 interface on two 75 ohm coaxial
cables direct to either the Operator Switch or an
Operator Digital Distribution Frame (DDF)
which is co-located with the Operator Switch.
-----------------------------------------------------------------------
Shouts to Night, Datawar, Substance, Zomba, Psyclone, Monty, Crypt,
grip, euk....
-----------------------------------------------------------------------
EOF
D4RKCYDE 1997,1998,1999,2000,2001
F41th Magazine http://www.f41th.com
#darkcyde Efnet.
mailto: hybrid@f41th.com.
