larry@kitty.UUCP.UUCP (04/11/87)
In a recent article kludge%gitpyr@gatech.gatech.EDU (Scott Dorsey) writes: > Is there any problem using telephone equipment which is not FCC licensed? > I have some surplus Army field telephones (modified for common battery use), > and some homebrew phones which are around the house. I certainly hope that > I can still operate it, although I could understand being prevented from > selling such stuff. I will give you two answers: (1) The OFFICIAL answer is that you are only allowed to connect devices to the telephone line which are FCC registered (which carry FCC registration numbers), or devices which were "grandfathered" at the time FCC Part 68 took effect. Grandfathered devices are listed by the FCC and by operating telephone companies, and are devices which were deemed acceptable to connect to the telephone line prior to the issuance of FCC registration numbers. Most of the grandfathered devices consisted of telephone answering machines, modems and PBX's; these devices would now be at least 11 years old, and would be pretty much obsolete. (2) The UNOFFICIAL answer is that you could damn well connect anything you want to the telephone network, and not have a problem PROVIDED that the device was a telephone with no external source of energy (i.e., not some 120 VAC powered device). Devices that utilize AC line power and are neither FCC-registered nor grandfathered should be carefully checked for powerline leakage before use. It is extremely difficult to cause physical harm to the telephone company cable plant or central office equipment, but it is possible if you intentionally try. It is also difficult to cause crosstalk on telephone company cables, but it is possible if you do something pretty stupid (like send a +20 dBm tone over the line), or if your motive is intentional interference. <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
larry@kitty.UUCP.UUCP (04/11/87)
In a recent article howard@cos.UUCP (Howard Berkowitz) writes: > One minor point (minor until it gets you): dial modems, > especially the higher speed ones, may not work properly > because the network interface device has resistors set > for voice use, not data. It is worth letting your telco > know you are running higher speed data, because your > network interface should then be configured to give you > better signal to noise ratios, not important for voice. > > If they're going to charge more for data access, do this > only after you have unacceptable problems. Your problems, > however, often are in the jack, not the local loop. The above article raises a good point, but it is not quite accurate. First, let's talk about data jacks. There are two types of data jacks with resistors: 1. The Fixed Loop Loss (FLL) jack, such as the RJ41S (RJ42S and RJ43S for A-lead control where data lines terminate in key equipment). This jack provides direct tip and ring access through pins 5 and 4, respectively. It provides an attenuated tip and ring access through pins 2 and 1, respectively. It is intended that voice signals utilize the direct line connection, while data signals utilize the attenuated line connection. The value of the attenuator pad will be set by the telephone company according to loop loss so that the effective loss (total of loop loss and pad loss) between the jack and the central office will be between 8 and 9 dB. The net result is that a 0 dBm data signal sent by the modem will reach the central office at -8 to -9 dBm. So the point is: the line connection through the resistor pad is for DATA use, and NOT for voice use. 2. The Programmable Loss jack, such as the RJ45S (RJ46S and RJ47S for A-lead control where data lines terminal in key equipment). This jack provides direct tip and ring access through pins 5 and 4, respectively. A resistor installed by the telephone company across pins 7 and 8 (leads PR and PC) tells the modem what output level to send. An infinite resistance tells the modem to send at -9 dBm; 19,800 ohms sends at -8 dBm; 9,200 ohms sends at -7 dbm; 5,490 ohms sends at -6 dBm; ...; and a direct short sends at 0 dBm. This resistor is sensed by internal modem circuitry, and has nothing to do with direct telephone line tip and ring. The telephone company will set the programming resistor such that the transmitted data signal will enter the central office at between -8 and -9 dBm (i.e., modem transmit level + loop loss = -9 dBm). Telling the telephone company that you are running "higher speed" data will most likely accomplish nothing. The telephone company could care less about your data rate or noise concerns (at least for the price of a POTS line); all the telephone company cares about is that your transmitted data signal reaches the central office at between -8 and -9 dBm. Period. Speaking candidly, if you feel compelled to send data at a greater transmit level, you can defeat any FLL or Programmable data jack and send at 0 dBm - regardless of actual central office loop loss. While I don't want to get off on a tangent here, the fact is that transmitting at a higher level is NOT necessarily going to result in a lower error rate. I can vouch for this from extensive personal experience. Most of the newer modems have receive threshholds of < -40 dBm; this is really quite a bit of sensitivity. Transmitting at a higher-than-necessary level can "strain" the ability of the band-pass filters in the modem to reject the locally-generated transmit signal, and leakage of the transmit signal into the modem receiver can often exacerbate a data line error situation. So the moral is: LOUDER is not always BETTER. :-) <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
randolph@cognito.UUCP (04/17/87)
At the end of a very good informative article on RJ-41 & RJ-45, Larry Lippman (larry@kitty.UUCP) writes: > While I don't want to get off on a tangent here, the fact is that >transmitting at a higher level is NOT necessarily going to result in a lower >error rate. I can vouch for this from extensive personal experience. >Most of the newer modems have receive threshholds of < -40 dBm; this is >really quite a bit of sensitivity. Transmitting at a higher-than-necessary >level can "strain" the ability of the band-pass filters in the modem to >reject the locally-generated transmit signal, and leakage of the transmit >signal into the modem receiver can often exacerbate a data line error >situation. > So the moral is: LOUDER is not always BETTER. :-) Moreover, some telco equipment does *not* have automatic gain control and cannot handle "hot" levels. This is especially a problem with older frequency-muliplexors -- the high levels cause overmodulation and the hot signal inteferes with adjacent signals on the same carrier. (I wonder: was this why telcos used to insist on Data Access Arrangements?) -- Randolph Fritz sun!randolph randolph@sun.com
larry@kitty.UUCP.UUCP (04/18/87)
In a recent article dave@lsuc.UUCP (David Sherman) writes: > I have two lines at home, one normally used for a modem. > > I'm curious to know whether I can construct something which will > let me achieve conference calling by using both lines -- the > electronic equivalent to holding the mouthpiece of one phone to > the earpiece of another, I guess. In other words, if I'm at home > talking to my wife on one line and our broker on the other, can > I do something which will let them hear each other as well as me? > > I realize I can get conference calling from the phone company; > I'm curious as to whether I can do it with what I already have. There are several methods of constructing a two-line conference device: 1. The _ideal_ method utilizes transformer isolation of each line, with a voice-switched hybrid-amplifier which supplies gain to the conference circuit (to compensate for connection loss so that the _distant_ parties can adequately hear _each_other_.) 2. A passive circuit employing transformer isolation. 3. Capacitor coupling of the two lines, possibly in conjunction with a resistor or inductor to "hold" one line (the local telephone set provides the DC resistance to hold the other line). 4. Direct hard-wire connection of the T (tip) and R (ring) leads of each line. Of the above, (1) is the best approach, but such a circuit is not trivial to design and build; (3) is a rather poor approach, and (4) is absolutely atrocious (although it does work, after a fashion). This leaves (2) which, if done using the proper components, works reasonably well, is inexpensive and easy to build, and is 100% safe to the telephone network. This article furnishes plans to build a conference device based upon that method. While building such a device and attaching it to the telephone network without it having been FCC-registered may be a technical violation of your "agreement" with your telephone company, people connect such non-registered devices every day, with no one being the wiser. Since the enclosed circuit is intrinsically "safe" (i.e., no ground reference or external power is used), I have no hesitation to provide its construction plans. Installation and operation is, of course, at the sole risk of the user. This device connects to the T and R leads of each line, through a four-pole switch. When the switch is operated, the lines are coupled together, and are also held through a resistor/inductor so that they do not disconnect from the central office. Actual operation is at the discretion of the reader, and should be self-evident. An example would be if two single-line telephones were present on a desk, with the device and its switch connected between them. In operation, a call would be made using telephone 1; the handset of telephone 1 would be put aside while telephone 2 was used to dial the other distant party; when the party on telephone 2 answered, the conference switch would be operated and telephone 2 hung up; the user would continue to use telephone 1 for the duration of the call, and would release the conference switch to drop the telephone line holds (actually on both lines) when the call or the conference was completed. The user could, of course, choose to talk with telephone 2 after the conference was completed, or could originate the call on telephone 2. The key to a successful implementation of this design is the selection of a proper coupling transformer (called repeating coil, in telephone parlance). Any 'ole audio transformer will NOT work, since each winding needs an EQUAL impedance between 500 and 1,000 ohms, IN ADDITION to being able to handle at least 50 milliamperes of DC current without saturating; it is this latter requirement which limits the transformer selection. The best transformer would be a "surplus" telephone repeating coil, with a type designation of 120C or 202A. Such a device can often be purchased for a few dollars from a surplus store or hamfest. Other sources would be a "friend" at a telephone company who would have access to their surplus equipment (such surplus abounds these days as telephone companies replace older equipment with ESS). This repeating coil has eight terminals since it has split windings; I have furnished the proper connections for these terminals (improper connection may result in very poor operation). If you can't find a surplus repeating coil, then try a regular audio transformer that can handle the DC current; some typical part numbers would be Stancor TA-52, TAPC-52; Triad TY-305P; UTC A-22, HA-108. Equivalents to these transformers can often be found surplus for a few dollars. If a transformer is used in place of the repeating coil, the transformer will no doubt have only one winding on each side. In this case, ignore any center-tap lead, and just connect the resistor-capacitor below in series with one lead. If anyone is wondering why the resistor-capacitor is connected _between_ the split windings below it is because, well, er, that is just the "traditional" way telephone circuits are designed when one has split windings. There is nothing special about the resistors or capacitors; just follow the specs below. Build the device in a case, with all leads properly insulated from each other and from the outside world. Switch REP Switch T (Line 1)______________X_____ | | _____X______________(Line 2) T X 2 ) | | ( 4 X ) | | ( 1 ) | | ( 3 _________) | | (_________ | | | | | | R1 / C1| | | C2 | R2/ \ _|_ | | _|_ \ / ___ | | ___ / \ | | | | \ / | | | | / |____|___ | | ____|____| 6 ) | | ( 8 ) | | ( 5 ) | | ( 7 R (Line 1)______________X_____) | | (_____X______________(Line 2) R X X Switch Switch REP Repeating coil (i.e., transformer), type 120C, 202A, or equivalent (see text) R1, R2 Resistor, 600 ohms, 2 watts C1, C2 Capacitor, non-polarized, 2.0 uF @ 100 WVDC Switch Four-pole single-throw toggle or rotary switch <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
larry@kitty.UUCP (04/19/87)
In a recent article Greg Earle (earle@jplpub1.JPL.NASA.GOV) writes: > I live in the 213 Area code in L.A. I recently moved, and want to get two > lines for my new abode, one of which I will use exclusively for a modem > line. When I talked to Pacific Bell I was told I could (for a nice high > fee, of course) get a `Data Access Line' which would (presumably) run from > the local switching office to my home; a higher grade line would replace the > normal voice grade phone line. I was told that this was recommended for > anyone doing data transmissions of 2400 baud or higher. I almost bit; but > then I thought, what about the rest of the way? I would be calling JPL in > Pasadena 99% of the time, which is in Area code 818, prefix 354. Since I'm > not a TELECOM expert, I just surmised that the calls I would make would go > from my home, over my `good' data line, to the local switching office; then > to whatever the local switching office for Pasadena is, and then over a > (presumably) standard voice grade line to my other modem. > > My question for you experts is (a) is this something like the real path that > the call will take (3 hops; home <=> switching office <-> s.office#2 <-> work) > and (b) if this is so, then is there any point in getting a higher grade line > for one's home, when one has no control over the line quality for the other > 2/3 of the connection ?!? Let's break up this discussion into two areas: (1) quality of central office subscriber lines (i.e., between your home/office and the telephone company central office); and (2) quality of lines between telephone company central offices. I'll answer (2) first, because it is the easier answer. In general, the quality of an interoffice trunk (i.e., a line connecting two telephone company central offices) is FAR superior to the quality of any subscriber line. In keeping with the DDD network operating goals and an overall transmission design plan called VNL (Via Net Loss), the transmission loss on most interoffice trunks originating at End Offices (Class 5) trunks is carefully kept below 4.0 dB. Interoffice trunk transmission loss on Toll Center (Class 4) and up to Regional Center (Class 1) switching offices is carefully kept below 2.6 dB, or even below 1.4 dB, depending upon the path. Such interoffice trunk design is generally done so precisely that loss is kept within +/- 0.1 dB of the design goal on any interoffice trunk of a given path to assure a uniform transmission quality. In addition, such interoffice trunks are generally equalized to have a reasonably flat transmission characteristic between 300 and 3,000 Hz. Furthermore since most interoffice trunks originating in End Offices (except in high-density urban areas where adjacent central offices are close together) are four-wire (i.e., separate receive and transmission paths - one for each direction) and are terminated in a precision hybrid-network, the transmission quality will be far superior to anything which could ever exist on a two-wire subscriber loop. All interoffice trunks of Toll Center and up origin are four-wire. Noise level, ERL (Echo Return Loss), and other parameters which affect the quality of transmission are also kept within precise design goals on interoffice trunks. The BOC's and larger independent operating telephone companies check the transmission quality of interoffice trunks on a regular basis, often using automatic test apparatus such as ATMS (Automatic Transmission Measuring System), CAROT (Centralized Automatic Reporting of Trunks), TFMS (Trunk and Facility Maintenance System), etc. Trunks which fail to pass these automatic tests are disabled until repair is effected. So the point is: under virtually all circumstances, you should have little concern about the transmission quality of interoffice trunks, as compared to your own subscriber loop. (IMPORTANT NOTE: The above applies to what is traditionally known as the DDD network; some of this standardization has gone to hell with the advent of Alternate Long Distance carrier. The above information should still be safely applicable if your call is intra-LATA in length, is inter-LATA but served by the same operating telephone company at both ends, or is routed through AT&T. This is NOT a "plug" for AT&T; it's just a simple fact of life since AT&T still runs all the major toll switching centers in the U.S.) Now we'll get back to the first topic, which is the local subscriber loop. Subscriber loops are generally designed based upon only two parameters: (1) DC resistance and (2) transmission loss at 1.0 KHz. Since most central office apparatus has subscriber loop resistance limits between 1,200 and 1,500 ohms, resistance of a subscriber loop is controlled to be within this range by selecting cable layout with sections that have large-enough wire gauges (the SMALLER gauge sections are generally CLOSEST to the central office). If the resistance limit still cannot be economically met with wire gauge selection alone, then a signaling range extender (loop extender) will be connected to the line; this device is always located in the central office. Under this condition, the subscriber loop resistance may be >> 1,000 ohms, but the loop extender has the sensitivity to support such a higher resistance. In simple terms, the transmission loss of a subscriber loop is directly proportional to its DC resistance - so a long loop will also have a large transmission loss. Invariably, subscriber loops greater than 10 kft (kilofeet) in length will have loading coils installed every X-kft (there are different loading schemes which use different spacing between loading coils); these loading coils add inductance which compensates for the attenuation of the loop that results from distributed capacitance. The worse case loss that any reasonable telephone company would impose on a subscriber loop is about 9 dB. New York Telephone, as an example, tries to keep loop loss to no more than 6 dB - but not every loop is that lucky. :-) If a maximum subscriber loop transmission loss goal of 6 to 9 dB cannot be met through loading and cable routing, then a voice-frequency repeater is installed on the subscriber line; this repeater is almost always installed in the central office on the required lines. Sometimes a combination loop extender-repeater is used, but in many cases there will be two discrete devices in the central office. Under most circumstances, subscriber loop loss is measured at only 1.0 kHz. However, a frequency-vs-attenuation plot of a subscriber loop can look like a roller coaster! Since the human ear is rather forgiving, for voice applications most telephone companies care little about the frequency-vs-attenuation curve on a POTS (Plain 'Ole Telephone Service) subscriber loop. However, MODEMS can care about this curve! Subscriber loops which run through mixed gauges of loaded cable, and/or run through voice-frequency repeaters (especially of the older E6 variety) can have some pretty ugly frequency-vs-attenuation curves. The only way to flatten the curve (and thereby make the line more attractive to data) is by means of an equalizer, a better quality repeater (hybrid rather than a negative impedance type like the E6), along with more careful design engineering of the particular loop. This equalizer, repeater change, and additional engineering is not necessary for most voice applications - so it isn't done. However, an equalizer and additional engineering DOES result in a superior subscriber loop for data purposes. So, telephone companies generally charge more money for a better subscriber loop design for data applications. If you are making serious use of data transmission at > 2,400 bps, the comparatively small additional monthly and installation charge is well worth it to get a better subscriber loop. I don't make a habit of defending telephone companies, but I must say that I feel such an additional charge is reasonable. They do have to install additional equipment (under most circumstances), and certainly do have to perform specific engineering on the design of the subscriber loop. As far as noise level on subscriber loops is concerned, this is generally caused by wet terminals and splices and is really a repair problem. There is little that can be done to reduce noise level on a subscriber loop other than to track down wet or poor splices. However, for the price of a better quality loop, one generally gets a quantitative noise measurement with some attempt at repair if the noise level is beyond normal limits. Now, to sum up and answer the $ 64 question: In my opinion, for data transmission > 2,400 bps on LONGER SUBSCRIBER LOOPS (say > 2 miles from the central office), an additional charge for a better quality loop (i.e., flatter frequency response and lower attenuation) IS a worthwhile expense. At least, with a known good loop of known characteristics, one can look elsewhere should data errors become a problem. <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
howard@cos.UUCP (Howard C. Berkowitz) (04/22/87)
I do want to emphasize that it's bad practice to transmit at a higher level than that which is designed for a local loop; it's quite accurate that louder is not always better -- but it is important to be loud enough. The worst horror story I've heard on excessive level, which may be apocryphal: An [unnamed by my informant in DATEC] user called his telephone test board for a problem with the remote site's Bell 829 data auxiliary set. [This is an interface device, for Telco use, which has a tone-operated loopback feature]. The user complained that the unit would not go into loopback. Now, 829's are for telco use, not customer. Most telcos are happy, however, to let customers use them for fault isolation. The 829 is put into loopback with at least a 1-second application of 2713 Hz tone. In the story, the helpful test board tried to loop back the 829 in question. It worked perfectly, but the user couldn't get it to loop. The test board then asked the customer to describe exactly what he was doing; the answer was "oh, I send out 2600 Hz at a good hot +10-20 dBm -- the limiter drops it to a legal level." Horrified silence from test board. A mystery of the last week just resolved. Over the past week or so, the telco had been installing a new electronic switch in its central office, a switch equipped with toll fraud detection. "Blue Box" fraud uses a 2600 Hz tone; somehow, the user assumed this was the loopback frequency. Because the user 2600 tone went out at an extremely high level, it crosstalked a large number of pairs in its cable. The new switch kept reporting massive simultaneous toll fraud attempts (i.e., pure 2600 on a subscriber loop), a sufficiently large number that equipment failure was assumed. The switch had been torn down repeatedly to find out why it assumed massive toll fraud was in process.
larry@kitty.UUCP (04/26/87)
> Finally, If both my best friend and myself set call forwarding to > each other, what happens when someone calls? > > [ ... Also, if you forward to someone who forwards to > you the call goes "click", "click", busy. The two clicks are the > trunks going between your central offices. If you do this to someone > on the same ESS machine, it gives you a busy immediately. This is > useful especially if you have your calls forwarded from home to work > and vice versa. You can safely forward your calls bi-directionned until > you get home and clear the forwarding. Also, I crashed our central office > forwarding once to someone on the same machine (that was 12 years ago). > --jsol] Re: the comment about "crashing" the central office... If anyone finds this hard to believe, I know of a specific example where there was a bug in a particular generic software release used on some early #2 ESS machines which had just been installed by New York Telephone to replace SxS CDO's during the mid 70's. By using three telephone lines, and setting them up to forward as A --> B --> C --> A, a call from another telephone to A as a "seed" would force the #2 ESS to crash and switch to the standby processor, whereupon a second occurence would wipe out the standby processor until the call attempt was discontinued. This situation would also set off remote alarms at the SCC which handled these unattended #2 ESS machines. Needless to say, a software patch was quickly developed by WECO... The above scenario was probably unanticipated by the WECO team that designed and wrote the software. The public, however, seems to have a knack for discovering these flaws rather quickly. <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
larry@kitty.UUCP (04/26/87)
While the Texas Instruments TCM1520A is a nice IC, it is possible
to build simple and reliable ringing detection circuits by other means.
Here are some suggestions which may be helpful in the design of circuits
which detect ringing:
1. When a telephone goes on-hook and off-hook during hookswitch
(i.e., line switch) operation, a voltage transient is generated whose
voltage is the same order of magnitude as a ringing signal. When a
rotary dial is used, each dial pulse is a momentary line open which
also generates these voltage transients.
A poorly designed ringing detector circuit will falsely detect the
above voltage transients as ringing signals.
To avoid this problem, ALL reliable ringing detector circuits
require a time constant. NO ringing detector circuit (unless it is has
frequency discrimination - which is extremely rare) can tell the difference
between on-hook/off-hook transients and the ringing signal itself based
upon a voltage threshhold ALONE.
Such a time constant can be established by three means: (1) integrating
the rectified voltage from the telephone line with a resistor-capacitor
before it drives an LED or relay; (2) using a thermistor in series with the
LED or relay (a traditional design approach, but the "right" thermistor is
difficult to obtain); (3) providing a specific timing circuit which looks
at the output of the optoisolator or relay, and requires that a signal be
present for a minimum period of time before asserting an output logic line.
A reasonable integration time constant is between 200 and 600
milliseconds; i.e., the ringing signal must be present for this time period
before a detection logic line is asserted.
2. All ringing detector circuits should have their telephone line
connection electrically isolated from ground, and should be coupled to the
telephone line using a series capacitor. In general, the value of this
series capacitor should not exceed 0.68 uF, and such a capacitor should
be rated at 200 WVDC. Excessive capacitance will cause voice-frequency
attenuation on the telephone circuit, and may also result in premature
"ring tripping" and dial-pulse distortion.
In general, the effective DC resistance of a ringing detector
circuit - EXCLUDING the capacitor - should be a minimum of 1,000 ohms.
Following the above capacitance and resistance constraints should
result in a ringing detector circuit which has a REN of less than 1.0 on
the "B" scale, and consequently should not interfere with proper operation
of the telephone line.
3. Optoisolators are nice for ringing circuit detection, but proper
and reliable ringing detector circuits can be made with relays. Use a
sensitive "plate" relay of 2,500 to 10,000 ohms resistance. Connect a
full-wave bridge rectifier to the telephone line using a series capacitor;
connect the DC output to the relay in series with a resistor, and place a
capacitor across the relay winding to provide an integration time constant
(be sure to have this capacitor rated at at least 100 WVDC!).
If your application is a ringing "extension" circuit - like to
drive an AC line horn, bell or light - you may find a plate relay with
a contact current rating sufficient for the job. This makes for a pretty
simple circuit.
Plate relays with the required resistance and sensitivity are
often available surplus for a couple of dollars. Do NOT use an AC relay
rated for 120 VAC; AC relays of this type generally do not have enough
sensitivity and a high enough resistance for telephone applications.
Also, note that some plate relays (like certain Sigma models) have their
body as the common contact - so these relays MUST be properly insulated
from the case and outside world.
The use of a relay to directly detect ringing and control an AC
power line circuit is a well-established design technique; however, use
extreme CAUTION when wiring such a circuit so that faulty construction
does not permit accidental connection between the telephone line and AC
power line!
4. If you are serious about designing telephone circuits, take the
time to study the operation of a telephone line using a storage scope
with differential inputs (i.e., one input for TIP, one input for RING -
NEVER ground either TIP or RING).
You will notice that -48 volts DC is ALWAYS present on the
telephone line, even during the actual ringing. The 20 Hz ringing
voltage is actually superimposed across the -48 volts DC; this is
referred to as "superimposed ringing". Superimposed ringing is done
to assure rapid operation of the "ring trip" relay in the central
office trunk circuit.
Generally, telephone ringing is 1 second on, and 3 seconds off
(i.e., the "silent interval").
If you are using PBX extensions to "play with" for telephone
circuit design, beware that their behavior may NOT be the same as
central office telephone lines. For example, some PBX's use 30 Hz
rather than 20 Hz; and some PBX's do not superimpose the ringing signal
on -48 volts DC in the same fashion as a central office. Also, PBX's
generally provide a "hotter" ringing signal than a central office
because your loop resistance to the PBX is generally << 100 ohms.
5. None of what I have said applies to party lines. You should
never attempt to design telephone circuits for connection to party lines.
Not only might you be detecting ringing for other parties, but improper
design or connection might also result in YOU getting billed for THEIR
telephone calls!
<> Larry Lippman @ Recognition Research Corp., Clarence, New York
<> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry
<> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/
<> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?" larry@kitty.UUCP (05/01/87)
In a recent article johnw@well.UUCP (John Winters) writes: > I was wondering, would anybody out there be able to write me a very > simple schematic for 5 watt amplifier which would amplify > my outgoing voice on the phone? > I talked to my local Bell guy and he said 5 watts would be allowed > (my grandmother can't hear at all, along with other people i know and > this would be handy) Hmmmm... 5 watts into 600 ohms; that would be sending into the telephone line at about + 37 dBm. Is that hot, or is that HOT? Not only would your grandmother hear your voice, but so would everyone else in your central office! I didn't post this reply to make fun of John Winters; I believe his statement that the "local Bell guy said 5 watts would be allowed". I just want to point out the INCREDIBLE ignorance which is sometimes displayed by telephone company sales and administrative personnel who deal with the public. Many people naively believe that ANYTHING said by ANYONE at the telephone company must be correct. Wrong. If you have a technical question - especially concerning data transmission - INSIST upon speaking with a person who is an engineer (common titles: "customer services engineer" or "facilities design engineer"). You might get some moaning and groaning from a salesperson about such a request to speak with an engineer, but as far as I am concerned, it is your right to speak to a technically competent person when it concerns a technical question about transmission or network interface specifications. This does not mean you should abuse such a demand by asking questions about how to connect or repair telephones, or how to design or build telephone equipment; your right to obtain technical information should be properly confined to transmission and network interface specifications ONLY. You might be referred to a specific AT&T or FCC technical reference; I consider such a referral to be a reasonable response to a request for information, PROVIDED you are given a specific publication number and where to obtain it. Concerning the original article, any person who has a hearing impairment must solve the problem at THEIR END. Amplifier handsets with an adjustable volume control are readily available from AT&T Information Systems and from other vendors of telephone apparatus. Prices range from $ 30.00 to $ 60.00 for such a handset. <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cakitnele atn
larry@kitty.UUCP (05/02/87)
> In a recent article AWalker@RED.RUTGERS.EDU (*Hobbit*) writes: [discussion about SF-tone detectors in central offices] > Isn't this a bit redundant in these CCIS-ridden days? I would think so! Most toll fraud today occurs through the fraudulent use of calling card numbers. However, during the 1970's when "blue box" fraud reached its peak, the Bell System in particular did use 2600 Hz SF tone detectors. One such device was called [somewhat euphemistically] a Multichannel Tone Test Unit (MTTU). One MTTU had the capacity to monitor up to 100 trunks. The MTTU could be used in a local office to monitor outgoing DDD access trunks, or in a tandem office to monitor 2-wire or 4-wire intertoll trunks. In the MTTU, each trunk connection had a dedicated SF tone receiver which would alarm if an SF signal longer than about 200 ms was detected. The sensitivity was pretty decent - something between -35 and -40 dBm - so it COULD conceivably be susceptible to the crosstalk situation mentioned in the earlier article. The MTTU had a trunk identification unit, which would send the identity of the "offending" trunk to the Call Identity Indexer of the central office CAMA or LAMA recording apparatus. This would allow the origin (i.e., calling number) of the fraudulent call to be ascertained. > Also, it seems rather improper for an office to assume that any occurrence of > 2600 on a subscriber loop indicates possible fraud. First of all, if someone > wanted to defraud he'd just hike down to the nearest pay phone. Second, there > are a lot of OCC switches that respond to 2600, so the phone co has another > think coming if they believe I'm committing toll fraud every time I clobber > one of them upon completion of a call. Fooey. If the [possibly apocryphal] crosstalk incident had occurred several years ago, I would believe it. If the incident is supposed to be contemporary, then I would be skeptical that an operating telephone company is still using such toll fraud detection apparatus (unless they have little or no CCIS and/or are still using CAMA trunks with local ANI - at least not likely today in an ESS office). Most message accounting today is LAMA; i.e., it is done in the local central office. Such message accounting has returned to the local central office primarily to permit message unit timing on local calls. So the point is: the LAMA knows every number that a subscriber has dialed (by dial-pulse and DTMF, that is). Assuming that there is no CCIS or 3700 Hz out-of-band signaling to cause a absolute denial of "blue box" usage, one can't implement a "blue box" fraud without gaining access to a toll switching office. And one generally can't gain access to a toll switching office without creating one of three situations: 1. Dialing an inward WATS number. Simple computer exception reporting from raw LAMA call data can ascertain if certain subscriber lines are making unusually large numbers of 800-number calls. Of course, such excessive usage can be perfectly legitimate, but detection of such high usage, along with other "anomalous calling patterns" can be used to pinpoint subscriber lines where toll fraud is suspected. A "roving" SF-tone detector could then be attached to _specific_ suspect subscriber lines. 2. Dialing directory assistance in other area codes. This is even easier to detect by exception reporting: one doesn't have many directory assistance calls lasting more than, say, three minutes! 3. Dialing an actual toll call, but applying SF before answer to reseize the toll switching office and dial a "more expensive" toll call. This is not very common because the subscriber line is still going to be billed for the original dialed toll call. In addition, most newer ESS offices closely monitor answer supervision on outgoing toll trunks; such monitoring makes it difficult to perpetrate "blue box" fraud. Failure to achieve answer supervision within say, three minutes results in a forced disconnect. Once answer supervision has been detected, its subsequent loss for more than say, 30 seconds will result in a forced disconnect. Furthermore, there is ESS software to monitor "anomalous" trunk answer supervision changes. > The user-end symptoms of 2600 detection seem to be as follows: Beeeep. Switch > disconnects your call, or whatever its fancy. Some switches drop the > connection to the office completely, forcing the call to throw back to the > office and return dial tone within a few seconds. At any rate, in the > background one can hear a small "grack" sort of click -- I would assume that > this indicates the bridging-in of the more sophisticated "fraud detection" > equipment that would listen for and report various other tones. This is > un-bridged again after about 20 seconds if nothing else happens. I could > determine this because in some offices the bridging equipment is flakey and > introduces extra line hum while it's connected. Good heavens! You actually tried it?! :-) :-) :-) You are most likely just hearing the originating register or its ESS equivalent being switched into the circuit to accept the anticipated MF signaling train, with the register "timing out" after 20 seconds of no signaling. Unless there is faulty apparatus, you will NEVER aurally detect the presence of any SF monitoring devices; all such devices use bridging amplifiers that result in an effective bridging loss of no more than 0.05 dB. > Would someone closer to the technical end of the above like to explain how > this works in greater depth? And what is generally done with the generated > reports when there's obviously no "fraud" happening on a given loop? Telephone company security personnel react very cautiously to any suspected "blue box" fraud. If an SF-tone detector or exception reporting software results in "hits" for a given line on several different days, chances are a dedicated SF-tone and MF signaling detector will be attached to the suspect subscriber line. Further information will then be obtained that can be used in a prosecution; mere detection of SF tones is insufficient. It is necessary not only to know exactly what destination number was dialed, but to have some idea as to the identity of the _person_ using a given subscriber line; merely knowing the subscriber line number where the fraud originates is insufficient - the identity of the actual _person_ making the call must be ascertained. In case anyone is wondering, it is the absolute right of any telephone company or communications common carrier to attach such monitoring apparatus to any subscriber's line. Furthermore, under most circumstances, it is also an absolute right of any such telephone company or communications common carrier to aurally monitor any subscriber line to detect fraud; this may be euphemistically referred to as "service observing" - and is more common than telephone companies would like their subscribers to believe. The point I am trying to make in the above is that "blue box" toll fraud is disappearing, and the use of toll fraud detection apparatus is consequently diminishing. While the incidence of "blue box" toll fraud has decreased, it has unfortunately been replaced by fraudulent use of calling card numbers, and most recently by cellular telephone "spoofing" fraud (which is probably the worst can of worms yet!). <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
jeh@pnet01.CTS.COM (Jamie Hanrahan) (05/03/87)
Here is an absurdly simple ring detector which I threw together years
ago, before hobbyists had access to optoisolators:
Scrounge an old phone with a conventional bell. Disconnect and throw
away (or put in the junk box) everything but the 425B network (or
equivalent), the ringer coil, and the external ringer capacitor (if
present). (The 425B network is the mysterious box with altogether
too many cryptically-labeled screw terminals on the top.) Place a
magnetic reed switch in close proximity to the ringer coil. Connect
the reed switch to whatever.
I never had any problem with this detecting dial pulses as rings, and
it is rather likely that it conforms to phone company specs.
--- Jamie Hanrahan
(uucp: {akgua | hplabs!hp-sdd | sdcsvax | nosc}!crash!pnet01!jeh)
(arpa: crash!pnet01!jeh@nosc)
(internet: jeh@pnet01.CTS.COM)larry@kitty.UUCP (05/04/87)
In a recent article steves@tektools.TEK.COM (steve shellans) writes: > In my home I have a touchtone phone. When I press a number, such as > 7, for example, I hear 7 clicks coming back at me. Even though > I can dial a complete number, including area code in a couple of > seconds, the wait after that while I listen to the entire 'readback' > is very annoying. (The number I dial most often is 790-0000, which > is the local number for Allnet -- it seems to take forever.) > > From phones at work there is none of this, and all (outside) calls > go through very quickly. I assume that you have a "true" touch-tone (DTMF) telephone at home; not one of these touch-tone dial units which really put out dial pulses. In either case, the following is still applicable to the situation of using a touch-tone telephone in some older central offices. It sounds like your home telephone service is furnished by a "progressive control" electromechanical central office, such as step-by-step (SxS) or Stromberg-Carlson XY, that has BEEN UPDATED FOR DTMF SERVICE. Since you mention later in your article that your telephone company is GTE, I would bet money on the office having Automatic Electric 35E-type SxS apparatus. In general terms, "progressive control" central offices route the call through a series of devices known as "selectors" - one digit at a time in REAL TIME as the number is dialed ("step-by-step", as they say :-) ). This type of central office was designed around the use of a rotary dial in the subscriber telephone. Since these are electromechanical switches, there are minimum times required for the switches to operate. The dialing rate was standardized to accommodate the electromechanical response time of the switches. The call-processing speed of such offices is therefore limited by rotary dial pulsing specifications, which are typically: 1. Pulsing rate is a nominal 10 pulses/sec, which can typically vary from 8 to 12 pulses/sec. A digit "1" has one pulse, a digit "2" has two pulses, ..., and a digit "0" has ten pulses. Therefore, a digit "0" typically requires 1.0 seconds to dial. There are further specifications for the pulses within each digit. Pulses are line opens, and are generally specified as being 60 milliseconds open followed by 40 milliseconds closed; also referred to as 60% break (since each pulse interval is 100 milliseconds). 2. Inter-digit dialing interval (i.e., between dialed digits), of at least 0.25 seconds. So, to dial four "0"'s will require a minimum of 5.0 seconds (4 dialing intervals + 4 inter-digit intervals). So doing a bit of calculation, at BEST, dialing your example number of 790-0000 is going to take at least 8.35 seconds before the call can reach the 790 central office. This assumes an interoffice trunk between your office and the 790 office - if not, add another second for local tandem switching. In addition, add another 1.0 second for call processing in the 790 office. Now as I said earlier, these SxS and similar central offices respond only to dial pulses, because the pulses themselves control the switching apparatus. There is no way that the switching apparatus per se can deal with DTMF signals. However, for a number of years there have been converter circuits for use in SxS offices which receive a string of DTMF digits, decode them, store them as digital information in a register, and then outpulse the digits as rotary dial pulses at the nominal rate of 10 pulses/sec. These converter circuits are installed between the "linefinder" and the "first selector". When a subscriber line goes off hook and requests dial tone, an idle linefinder is selected by some simple relay arbitration logic. Each linefinder is a type of SxS switch which is dedicated to one particular first selector; when the linefinder completes its job and connects the tip and ring of the subscriber line to the first selector, the first selector returns dial tone. At this point, the first selector can respond in real time to the first dialed digit. Subscriber lines are typically arranged in groups of 100 or 200 lines, and each such group has access to a maximum of 10 or 20 linefinder- first selectors. The actual concentration ratio depends upon the particular type of SxS apparatus and the traffic design of the central office, but the typical concentration ratio at this point is 10:1. Typically, only ten percent of all lines in a given line group can make an outgoing call; if all linefinders in a given line group are busy, a line requesting service waits for for dial tone - since the dial tone is the indication that a first selector is available. So the point is: these DTMF converter circuits get installed before the first selector, are shared by a number of other lines in the same group, and in effect "fool" the first selector and all subsequent switches into believing that they are being controlled by a rotary dial. Since one can enter DTMF digits in a keypad at least 10 times faster than using a rotary dial, and since the SxS or other progressive control central office can only operate as fast as a rotary dial - use of the DTMF converter results in no faster switching time than using a rotary dial. The "clicks" you hear correspond to the dial pulses being generated by the DTMF converter, and are the impulse noise created by the operation of the selectors in your dialing path. If there is no faster switching time by adding DTMF converters to a SxS office, then why do telephone companies install such devices? Because customers "feel better" knowing that touch-tone is now available in their particular central office, because the telephone company can charge more for the touch-tone service, and because telephone companies don't want to spend the money to replace the SxS office with ESS! > Whenever I travel on business and need > to make calls, I always find electronic switching. You might still be going through a #1 or #5 crossbar office, or some other common-control electromechanical office; such offices when equipped with DTMF originating registers will usually switch a call in a short enough period of time as to be indistinguishable from ESS. > My question is this -- how unusual (in the U.S.) is the kind of > switching that I have from my home phone. If this is something > pretty rare, I would like to contact my phone company (GTE) and > the state utilities regulator to bring some pressure to bear to > update the equipment into the modern world. I don't think there is much pressure that can be brought to compel an operating telephone company to upgrade if they don't want to. They've given you DTMF service, they've given you DDD access, they've probably implemented coin-free operator, 800 and 911 service at coin telephones (done with a similar adapter card installed in the coin telephone linefinder group), and that's really about all the telephone service that any operating telephone company is OBLIGATED to provide. ESS features are nice, but they cost money to provide since they require total replacement of the SxS office with ESS equipment. New ESS equipment for a central office typically costs between $ 500.00 and $ 1,000.00 per subscriber line, with the higher figure being more common. That's a lot of money - that the telephone subscriber has to eventually pay for. This means that SxS and other electromechanical offices in "outlying" areas will be around for years to come. I don't disagree with your desire for better quality telephone service, but the point is: who is going to pay for the ESS to provide it? > Also, does anyone know when the heyday of this kind of equipment was? SxS and other progressive control central office equipment for smaller central offices peaked in usage during the late 1960's; common control electromechanical switches such as crossbar and its predecessor panel (ugh!) were just not economical for offices with less than say, 3000 lines. During the 1960's small "packaged" crossbar switches became available (like those from NEC), along with hybrid ESS-electromechanical switches, followed by true ESS switches. These switches made it economically feasible to replace SxS and similar progressive control central offices. <> Larry Lippman @ Recognition Research Corp., Clarence, New York <> UUCP: {allegra|ames|boulder|decvax|rocksanne|watmath}!sunybcs!kitty!larry <> VOICE: 716/688-1231 {hplabs|ihnp4|mtune|seismo|utzoo}!/ <> FAX: 716/741-9635 {G1,G2,G3 modes} "Have you hugged your cat today?"
howard@sundc.UUCP (05/05/87)
In an earlier article, I referred to an incident, which I suggested possibly was apocryphal, of problems due to 2600 Hz crosstalk affecting tone fraud detection equipment. I was told of the several years ago, at an ANSI standards meeting by a telephone industry employee. The incident probably took place in the late 70's or early 80's, and possibly was in Chicago.