Internet Voice Button
Q: What is Nortel's Internet Voice
Button?
A: It is a value-added web tool, available
through Internet Service Providers, that allows a Web-browsing
consumer to call a company directly from its Web site while
keeping his or her Internet session active. In addition to
communicating by voice in real-time, the business representative
and consumer may push Web pages to each other and engage in text
chat.
Q: How do my customers call me from my Web
site?
A: They simply click on a Voice Button placed
on your Web page and within seconds their call to your business is
established.
Q: How does the service work?
A: A business incorporates a Voice Button
link on its Web site that, when clicked, passes information about
the customer to the Voice Button server operated by the Service
Provider. First time users are presented with a simple
configuration screen, where they indicate whether they'd like to
use a regular phone, Internet phone or request delayed connection.
The parameters entered become the default settings for future
uses, whether on this site or any other that incorporates the
technology. The user may update settings at any time.
The Voice Button server then establishes a
call to the customer using the method selected in the
configuration. When the customer answers this call, the server
initiates a second call to the business. The business and customer
are then connected.
Q: Are my customers using a phone when making
their calls?
A: If your customer has two phone lines to
provide telephone and modem connections, he may choose to
establish the call via regular phone. Once Voice Button is
clicked, the phone will ring within seconds. If only one phone
line is available to provide both voice and Internet connections,
your customer has two options:
Internet phone - Placing the call using VoIP
requires the person to have a multimedia PC with headset (or
speakers and mic) plus Internet phone software such as Microsoft
NetMeeting.
"Call me in X minutes" - Modem users with a
single phone line may specify when their call will be placed
(within a few seconds to a few minutes) and disconnect their
current Internet session enabling the call to be taken via
phone.
Both the regular and Internet phone options
allow the customer to maintain an Internet connection while
speaking with the business.
Q: Do I, the business, have the choice of
taking calls on our existing phone system or via computer using
VoIP?
A: Voice Button currently allows for the
business to take calls by phone only. This makes the service as
seamless as possible to the business, requiring no new hardware or
software. Just take calls as usual.
Q: How good is the quality of
VoIP?
A: The sound quality of VoIP calls continues
to improve rapidly with new technology. However, the overall voice
quality depends upon many factors including the user's PC and
network connection.
Meridian HomeOffice II
Q: What is required on the PBX for Meridian HomeOffice II?
A: Meridian HomeOffice II requires an IPE module, digital trunking
and support for Extended Digital Line Cards (XDLCs).
Q: What software release is required for the PBX?
A: Meridian HomeOffice II is compatible with Meridian 1 Release 17
and above and SL-100 BCS 32 and above.
Q: What are the PBX platforms for Meridian HomeOffice II?
A: Meridian HomeOffice II can be used with the following Meridian 1
or SL-100 PBXs:
*Meridian 1, Options 11C, 21E, 51C, 61C, 71C, and 81C
*SL-100, Options 111 and 211
*Older systems that have been upgraded with IPE modules
Q: What application(s) does Meridian HomeOffice II support?
A: Meridian HomeOffice II is designed for telecommuting, one of the
three most common remote access applications (nomadic, telecommuting and
LAN-to-LAN) in the enterprise space. Meridian HomeOffice II provides the
following:
*Digital voice connectivity extended from the Meridian 1 PBX, with all
features and functionality
*A LAN Ethernet connection into the corporate data network
Q: Can Meridian HomeOffice II place data network calls to multiple
destinations?
A: Yes, Meridian HomeOffice II supports up to 32 user-specified locations,
where each location is a distinct network. On the telephone network the
Meridian HomeOffice II system is limited to on PBX connections per user.
Q: Can Meridian HomeOffice II place simultaneous data calls to different
destinations?
A: Yes, but only two, and digital telephone must be in offline mode
to allow this.
The two B-channels available with ISDN provide a maximum of two simultaneous
connections. Meridian HomeOffice II is designed to support one data
connection into a host environment when the digital set is active (that is,
online to the PBX). Since the digital telephone uses one B-channel when in
active mode, only one B-channel is available for a data connection. With
the digital set in local mode (that is, offline from the PBX), two simultaneous
data calls can be connected.
Q: Does Meridian HomeOffice II have analog phone interfaces?
A: Yes. The HomeOffice Router has one analog jack interface (port)
which can be used to support analog devices (such as a fax machine) at the
home office.
Q: Does Meridian HomeOffice II accept analog modem calls?
A: No. The HomeOffice Router does not have the necessary digital modems
to support this type of functionality. However, you can connect a modem
to the Fax port on the back of the unit to send and receive data and faxes
from a PC. This would be identical to using the modem on a regular analog
line.
Q: How do I attach my PC and digital telephone to the HomeOffice Router?
A: The Meridian HomeOffice II product is a bridge/router device
that provides an Ethernet network connection. To connect your PC to
the Meridian HomeOffice II product, install an Ethernet network
interface card (NIC) and load the appropriate drivers on your PC.
Use the Ethernet crossover cable (included) to connect the Ethernet
card in your PC to the Meridian's Ethernet port. This provides a LAN
connection between your PC and the Meridian HomeOffice II product.
The digital telephone is connected to the bridge/router device directly
to the RJ11 port labled MERIDIAN on the rear of the product.
Q: How many analog devices (modems, fax machines, telephones, etc.)
can be connected to the POTS (FAX) jack on the HomeOffice Router?
A: Any combination of devices that add up to a total of three
Ringing Equivalency Number (REN), which usually equates to about
three or four devices. Each of these devices acts as an extension off
the phone jack. Only two devices and two B-channels can be active at
the same time.
The greater the wiring distance, the fewer the extensions (exact
specifications depend upon cable type, specification, and so on).
Also, the greater the REN of the analog devices, the fewer devices the
router can support off the port.
Q: What kind of devices can be connected to the HomeOffice Router?
A: The Router offers three different communications capabilities to
the telecommuter: LAN Ethernet data connection, digital telephone set
extension and the use of an analog device(s) connected to the telephone
jack.
Q: Is bridging or routing a better solution?
A: Bridging or routing depends on your network configuration. As a
rule, you must route between dissimilar network addressed LANs, and
bridge between similar ones. The HomeOffice Router is especially suited
for routing from a remote location into a LAN backbone. Obviously, not
all network designs are the same. Talk with your network manager to
determine your particular needs.
Q: Is there an easy way to configure Meridian HomeOffice II?
A: Yes, there is. The Meridian HomeOffice II is user installable
versus technician installable. The telecommuter can use the friendly
installation wizard, along with auto-SPID detection, to easily configure
it The graphical user interface allows configuration in a point-and-click
environment.
Q: What channel is used for voice calls, the B- or D-channel?
A: There are three types of voice calls that can be placed on Meridian
HomeOffice II, whether it is digital voice or analog voice. They are:
* The voice call made while the digital telephone set is "online" to the PBX
* The voice call made while the digital telephone set is "offline" to the PBX
* The voice call made via the FAX port on the HomeOffice Router.
In any of these scenarios, the call is placed across one of the B-channels.
The D-channel is used only for ISDN BRI B-channel call setup and tear down.
Nortel Networks authorized distributor
channels can contact 1-800-4 Nortel for further assistance with
questions. For more information visit
www.nortel.com.
Reprinted with permission from Telecom
Reseller magazine
Nortel Notes
by Phil
Ruffin
Phones
Q: I need to add DID numbers to my Nortel PBX but all of the available
numbers from my provider conflict with my dial plan. Can I modify the incoming
numbers so they terminate on internal numbers that are different from
the dialed numbers?
A: Yes, you can use IDC (Incoming Digit Conversion) to change the numbers.
It will be confusing (at least it is to me!), but you can do it. In LD 49,
establish an IDC table with the incoming number first on each line,
followed by the internal extension number where you want to terminate the
call. Then in LD 16, change IDC for the route to YES, and add the IDC table
number at the DCNO prompt (for day mode) and NDNO (for night mode).
Understand though, that this may make you pull your hair out when you forget
it's in place and you're trying to figure out why things don't work the
way they should.
A better answer is to renumber whatever is necessary to allow available numbers
to terminate on your PBX with matching internal numbers. Maybe you can
renumber your trunk access codes or Park codes, or even some of the internal
extension numbers. Print a DNB list (LD 22) and review the conflicting
ranges of numbers to find the best match.
Q: I have three
lines on my 2616 set, but calls don't come in on the third
one. If the first two lines are busy, a third call will always
go directly to voicemail. How can I get it to work?
A: Look at the LHK (Last Hunt Key) prompt
in the station programming. You want it set to "2". While
you're at it, set LPK (Last automatic line Preference Key) to
"2" as well. This will allow the third line to be
automatically selected, so you can answer it by lifting the
receiver, if you have IRA (Incoming Ringing line preference
Allowed) selected in the CLS (CLass of Service).
Q: My 2616CT cordless phone keeps losing
its mind. My vendor came and reset the phone by removing the
battery and unplugging the phone from power and the PBX for a
few minutes. They said that's all they can do - it just has
problems sometimes. Is there any way to fix it?
A: This problem sounds familiar. Look at
the bottom of the set. Find the barcode label, and look for
"Rls # 1" or "Rls # 2". Yours sounds like release 1. Inform
your vendor that there is a free replacement program for the
first release of the 2616CT sets. Tell your vendor to refer to
Product Bulletin 2000-005 and "KPD # R99-H-34". That should be
enough information to get results.
Q: I changed the base cord on my 2112 set
to be able to move my phone farther from the jack, but now my
speaker doesn't work. When I change the cord back, it works.
Does my phone only work with a short cord? That doesn't make
sense!
A: You are correct - that doesn't make
sense. I suspect you will find the problem in the number of
conductors in the cord, not its length. The 2112 set requires
a special transformer and a six-conductor base cord to supply
voltage to drive the speakerphone. Your new cord probably has
only four conductors.
Q: Sometimes when I plug a digital phone
into the wrong jack, it will make all the indicators flash
together. Other times, it does nothing. Does this mean
anything?
A: Yes, it means you are careless about
where you plug your phones. You will be pleased to know,
though, that the flashing indicators indicate that you are
plugged into an analog port. This isn't 100% reliable, though.
You should test with an analog phone before you declare it an
analog port. I've seen defective digital ports with the same
indication.
Q: Whenever I dial 181 on a phone it
makes a loud beep and goes dead. The programming for the phone
completely goes away. The only way I can get it to work again
is to reprogram the entire phone. What can I do to make this
stop?
A: I suppose you didn't think of this:
sop dialing 181. No, that's too easy. It sounds like
you've found a feature called ASR (Automatic Set Relocation).
Actually, you only found the first half of the feature. The
second half works like this: once you have ports prepared to
accept the phones that have been "removed" from software using
the ASR feature, you can plug one of these "removed" phones
into the port, and it will magically reappear, reprogrammed on
that port. Isn't that a neat feature? ASR allows you to move
phones from port to port, and performs the basic programming
changes for you. The feature code is SPRE (Special PREfix) +
81 + ASR Security Code. You seem to have a SPRE of 1, and no
ASR Security Code.
You can make it much less likely that
phones will keep disappearing from software if you program an
ASR Security Code in the CDB (Customer Data Block). The prompt
is SRCD (Set Relocation Security Code).
I suggest you use a combination of digits
that will make it most unlikely for people to dial it
accidentally. Some people use four zeros, since it would mean
people would have to dial 1810000 to evoke the feature. Choose
a set of digits that you can remember if you want to use the
feature properly, and program it in LD 15.
Now that you have a grasp of how to use
this feature, you need to know that each time you reprogrammed
a phone instead of moving it by completing the ASR process, a
set of programming was stored in memory, and it may still be
there. Here's how to find it and remove it.
In LD 21, PRT the SRDT. This should give
you a list of all the sets stored, waiting for them to be
plugged into an available port and activated.
Next, use LD 50 to OUT the MTRT, followed
by the old TN. Do this for each of the sets listed above in LD
21.
Is that powerful? You bet. You can move
sets on the spot, without using a terminal at all. You still
may need to follow up with changes in DES and any
documentation you use, though.
Let me give you a little warning about
this feature. Some people have experienced what they believe
is memory corruption from using the feature. One story I heard
came from a user who was moving a long list of phones using
ASR. Just as the last phone was removed, the PBX initialized
and all the programming was lost. The phones had to be
restored by reprogramming them individually. I don't know
about that being a memory corruption, though - I think he ran
out of RAM in the PBX.
I haven't had an experience like that,
but I would strongly recommend printing out all of the phones
before relocating a long list, no matter what method you use.
And if you do suspect a memory corruption in telephone
programming, I suggest you LD 1. That will automatically run a
routine to inspect the telephone configurations and (in some
cases) correct memory corruptions that occur.
Q: We have a mix of 2008 sets without
displays and 2616 sets with displays. Often, when a person
leaves the company, his 2616 set disappears the same day. Is
there an easy way to keep track of these?
A: Yes, you probably find the set when
someone calls to have his display fixed. I usually check the
programming first, to see if it's a 2008. If so, I go and
confiscate the phone. I inform the user that he can have a
2616 set if he can get approval for the purchase, and I even
provide the purchase request documents. The original 2008 will
mysteriously show up within a day or two.
Q: Is it true I can use my phone to
enable my TTY port?
A: Yes. Your phone must have MTA in the
Class of Service. Use Key 0, and dial SPRE plus feature code
91. Using the telephone keypad, key in "LD#37##" and wait for
the overlay to load. Key in "STAT#TTY##". Your TTY should be
in the list, and it should show to be disabled. Note the TTY
number, and key in "ENL#TTY#" (key in the TTY number here)
"##". Check to see that the status changed by using
"STAT#TTY##" again. Exit maintenance mode by using "****".
Q: I have several phones that are
restricted from dialing local calls, but I need to allow them
to dial 911. How can I do that?
A: I think this is a question many
administrators and managers will (and should) be asking in the
near future, with states and municipalities beginning to
require 911 call capability from any public phone at business
sites, and the answer is not necessarily an easy one. There
are a couple of ways to do this, depending on the strategy
already in place for the current restrictions, and the release
of software available to you. Two dialing plans are common for
911 calls at businesses, and the lawyers have already chosen
one of these for some areas. Before I get into the
programming, let's talk about these two dialing plans, and you
can decide which one to implement.
The choice of lawmakers in some areas
(some say the FCC will require this eventually) has been to
make the emergency call as easy as possible by allowing the
user to simply pick up the phone and dial 911 to get emergency
services, just like at home. That sounds easy, doesn't it? I
have to admit that I was taken in by this at first, too. It's
compelling to make it so easy anyone can just dial 911 any
time from any phone without having to think about how to dial
special codes, like an access code to get an outside line. And
it can be easy to program in some cases. The trouble comes
when the local police start to get upset at all of the calls
placed to 911 in error. I discovered this problem myself when
I implemented this strategy some years ago on an Option 61.
Things sailed along just fine for a few days, and then people
started making mistakes dialing calls. There was the manager
who dialed 9-11+ instead of 9-011+ for an international call.
Twice. Then there was the conference phone that sometimes
dialed double digits. Several other calls were made to 911,
and each time the emergency equipment arrived, and there was
no emergency. Once, even I was discovered to be the culprit,
testing a terminal program installation that wasn't configured
properly. The police department got really irritated with us,
and wanted to put a stop to all of these false emergencies.
By now, you may have figured out that
this dial 911 only approach is not my favorite. While it is
required in some areas, I choose not to implement it until the
lawyers drag me into the switch room and give me a copy of the
requirements. Until then, I program systems to dial the way
most business workers expect. Dial 9 (or whatever local access
code you normally use), then 911. "But wait!" some people
point out to me. "What about the visitor in your building who
doesn't know how you use your phones, but has an emergency and
tries to dial 911?" I don't know how to answer questions like
that. Maybe you need to give people 911 instructions when you
issue a visitor badge. I still think most business people are
more likely to dial 9-911 in an emergency at work than to dial
911. Ask your people, if you want to find out what will work
best in your environment. Check your local requirements, too,
to see whether the lawyers have already decided for you.
Some older systems may have the phones
restricted by assigning a CLS (Class of Service) of FRE (Fully
Restricted), SRE (Semi-Restricted), or some other that will
restrict calls by type of connection (meaning station or type
of trunk). For these systems, especially if you have software
that is release 22 or older, you really should consider
implementing BARS (Basic Automatic Route Selection) in order
to properly allow 911 calls. It is a very flexible system and
can be programmed several ways to accomplish the same result,
so many people are confused by it. If reading the manual
doesn't help, contact your vendor for assistance.
Also, a very capable gentleman who goes
by the name GHTROUT has a great description on his web site.
Go to www.ghtrout.com and click on the BARS 101 link.
Beginning in release 23 (I think it was
23.55), a new way of programming this was introduced. With the
new ESA (Emergency Services Access) feature, the PBX allows
you to designate an emergency number for the system, and it
gives that number priority over the entire dialing plan. The
idea was to allow your users to dial 911 (or whatever
emergency number your community uses) without any regard at
all for the way the rest of the phone system is programmed. So
the emergency number simply overrides all other routing or
number plan information in your system. Pretty neat, I think.
It also gives you a way to notify a guard or attendant when a
911 call is placed, and even prints out details on the 911
call on the system terminal. I really liked this feature,
until I tried to program it so that calls could be dialed
9-911. It didn't work. Finally, someone pointed out to me that
the book specifically states that you must have CAMA
(Centralized Automatic Message Accounting) trunks for this to
work with PRI (Primary Rate Interface) trunks. But it should
work fine with all other trunks, including CO (Central Office)
Ground Start Trunks. I haven't had the opportunity to check
later versions of this feature to see whether Nortel has fixed
the problem for PRI trunks. I expect it to work in later
releases.
So, what will you do with this? Will you
reprogram your system to use BARS to handle these calls? Will
you use ESA to give priority to the calls? I invite all my
readers to let me know what you're doing with this, and what
trouble you find as a result.
Q: The 911 dialing information was helpful but I need more information
for our environment. We have several locations tied together using tie lines
and T1s. Some locations share a PBX and phone lines.
A: One way to handle 911 calls in a shared PBX is to use Tenant Services
software (also called Multi-Tenant software). Multi-Customer software
should also work. You can direct the 911 calls to dedicated local trunks
(if your hardware will support them) so the call will carry the local
phone number. If you need CAMA trunks to carry even more information, I
think you will need to have CAMA to each PSAP (911 office). As a last resort
you can use an external box that will provide the PSAP with the information
it needs, but it requires more ongoing administration.
In cases where you have PBXs tied together, the solution is easier, since
that PBX can send such calls out on local trunks with few programming changes.
I think that this would be a fun project to work on.
Q: What amplified handsets are compatible
with the 2008 and 2616?
A: You probably don't need an amplified
handset with those phones. You can use the built-in
amplifiers. While talking on the handset, increase the volume
by repeatedly pressing the right side of the volume bar at the
bottom of the face of the set. I haven't needed any additional
amplification for anyone yet.
Q: One 2616 set in our office doesn't
work as a speakerphone. I swapped phones, so that isn't it.
The speaker works, but the person on the other end can't hear
me when I test.
A: This should be an easy one. You
probably just need HFA (HandsFree Allowed) in the CLS (CLass
of Service) instead of HFD (HandsFree Denied). If you have a
feature programmed on key 15, it will go away. Key 15 is
changed to Handsfree/Mute when you activate the Handsfree
feature, just as key 7 is changed to Program when you activate
the display.
Features
Q: What can I do to
allow Last Number Redial on the phones where I don't have a spare
key to use?
A: On all the digital sets except the 2317 or
3000, you can use the feature without assigning a key. Just add
LNA (Last Number redial Allowed) to the CLS (CLass of Service),
and set LNRS (Last Number Redial Size) to the appropriate number
of digits, and the feature is programmed. To use this press a line
key twice, or lift the receiver and press the active line
key.
Q: All our phones sound alike when they ring.
How can I change the ringers so they have different sounds?
A: In the CLS (CLass of Service), you
probably have DRG1 (Distinctive RinGing 1) on all the sets. This
is the default - a high frequency, high-speed oscillation. You can
change the programming to DRG2, DRG3, or DRG4 to apply a different
ring to the phone. I usually set up a demonstration area with the
4 rings clearly identified so users can determine which ring they
want, then have them email me with their preferences.
Q: I have a group of phones set up with a MCR
(Multiple Call Ringing) key to share calls for software support.
Still, callers sometimes get busy signals. Shouldn't it always be
available for calls until the keys are busy on all the
phones?
A: Yes, this is a problem. It really doesn't
work like ACD (Automatic Call Distribution). I don't know a good
solution for this, since it should work, but doesn't. Have you
considered using real ACD? It gives you better control, plus
reports.
Q: I can't get the Hot Line feature to work.
I'm programming key #8 with "7 HOT D 4545".
A: You're making a common mistake. After the
key number "7" and "HOT D", you must insert the number of digits
that will follow before you enter the digits themselves. There are
four digits in the number 4545, so that would make your command,
"7 HOT D 4 4545".
Q: Is it possible to program a different
function in the place of the Handsfree Mute key? I get an error
whenever I try to program it. It happens when I try to replace the
Program key, too.
A: Yes, it is possible, but do you really
want to? The Handsfree Mute key is a requirement if the phone is
to be used as a speakerphone. To eliminate that function, use HFD
(HandsFree Denied) in the CLS (CLass of Service) instead of HFA
(HandsFree Allowed). After you make that change, you will be
allowed to program a different function on that key.
In the same way,
the Program key is necessary if you have a display on the set.
You can program NDD (No Digit Display) in the CLS to eliminate
most of the display functions, even if you have a physical
display on the phone. Q: What is MARP, and why would I want to
activate it?
A: I've heard this
question many times, and it amazes me sometimes that this
feature is still optional. I frequently find systems with MARP
(Multiple Appearance Redirection Prime) not activated, but I
can't think of a single reason NOT to activate it when the
system is first installed. MARP only
applies to situations where one extension appears on more than
one phone. This is commonly called a Multiple Appearance DN
(Directory Number). You might find this arrangement on a
boss's phone and his secretary's phone when the boss wants the
secretary to answer calls for him while he's busy (or at the
golf course). This is a useful strategy for some
manager/secretary situations where the secretary needs to have
full control over the boss's calls, or where the secretary
just needs to know when the boss is on the phone. Before MARP, it was a puzzle to figure out
which phone would have control over the behavior of the shared
DN. That's why MARP was invented - to make it easier to be
certain what phone controls each DN. Consider this example. The boss and
secretary share extension 3000, and the boss gives the
extension to his close friends, and even his wife. Today, the
secretary is out, and the boss is working in a conference
room, where he has the budget spread out all over the table.
He forwards his phone to the conference room, and plans to
answer his calls there. Uncertain of this technology, he even
tests this arrangement by calling his extension to ensure it
rings on the conference room phone. Only later will he learn
that his shared extension 3000 does not forward with the rest
of his numbers. Why? His phone is not in control of that DN on
his phone. Do you see what a mess this can become in a system
that doesn't have MARP activated? Without MARP, only math and tealeaves will
show you which phone is in control of a shared extension. Be
prepared to list all appearances of a shared extension and
look for the lowest numbered TN (Terminal Number) and the most
recent date. Be aware of whether or not the DN is included in
the Short Hunt (Last Hunt Key or lower). And expect to be
required to deal with this every time you make a change on any
of the phones with the shared DN. When
you have MARP activated, this is a simple matter. The first
phone programmed with the extension is the default MARP, the
one in control of the extension number. When you program an
additional appearance, the system will ask you whether to make
this appearance the MARP, or controlling extension. If you
just return, it assumes you responded NO. If you answer YES,
the new phone will control the extension. "What do you mean by control?" you may
ask. By control, I mean forwarding to another location (as in
the example), or hunting (busy), or FDN forwarding (no
answer). Before MARP came along,
companies would sometimes resort to programming analog phones
with shared DNs, in order to control call handling. This
helped clear some of the confusion, but led to higher cost
from using more TNs (Terminal Numbers), and made software
changes more complex. Here's a word of
warning, though. If you have a large system that does not have
MARP activated, don't just turn MARP on and then leave on
vacation. It's likely that some of the shared extensions will
not transition properly, and you'll need to reprogram MARP on
those. Otherwise, I suggest you always
use MARP from the time the system is installed. If the boss
knew how much time you will save by using MARP, he would thank
you (when he gets back from the golf course). Q: I get complaints
about how hard it is to set up a conference on the
speakerphones. How can I make it easier, so they don't
disconnect people?
A: Oh, yes, the
analog conference phones. What would we do without them? Not
that we wouldn't like to try. I have a
trick I've used several times that users like. If you can associate a secretary (or
someone else who can follow instructions) with the conference
phone, put its DN on her phone. The secretary can use that key
to establish a conference, and then have someone in the
conference room pick up the call on that phone. She gets to
use the conference key to establish the conference, so it's
much easier. Another advantage to using
this method is when someone needs to be added to the
conference after the meeting starts. The party will usually
call the secretary, wanting to join, and the secretary can use
the Call Join feature to put them together. She (or he) will
press Conference, press the conference phone line key, then
press Conference again. She can then hang up and allow the
conference to continue. It's pretty slick. When you first try
it, though, you may have a hard time hanging up on the
conference, for fear that it will disconnect them. But as long
as the conference phone has a COS (Class Of Service) setting
that allows the connections you made, the call will continue
on the speakerphone with no trouble. Q: Why doesn't my Nortel phone have a Drop
key like Lucent phones have?
A:
You're correct that the Nortel phones don't have a Drop key.
There are a couple of ways you can duplicate the function,
though, and even improve on it. If you
have release 23 or later, you can drop one party (conferee?)
from your conference by using the Conferee Selectable Display
key (on a 39XX phone it's labeled "ConfSelDsp"). It actually
works better than the Drop key, because you can drop ANY party
from the conference, not just the last one added. You scroll
through the list (you have to have a display to do this) and
use the display information to select the one to drop. You can also use the No-Hold Conference
key instead of the regular AO6 Conference key. The No-Hold
Conference key works like the Conference key you're used to,
except it doesn't "set aside" or "put on temporary hold" the
other party or parties while you add another one. The existing
conference hears everything while you add another party. As a
result, you don't have to press Conference the second time to
complete the conference until you're sure you don't want to
Drop that party, but everyone can talk and hear each other. I
personally don't like the No-Hold Conference key for myself,
because it doesn't give me the privacy I want, so I can see
whether I really want to add this new party to the
conference. Q:
My system won't let me delete a secretary's phone, and I think
it's because of something called BFS. What are BFS keys?
A: BFS (Busy Forward Status) keys can
be pretty useful, and sometimes frustrating. In a way, they
operate like DSS keys on a key system. A single key allows you
to see when a phone is busy, and you can transfer calls to
that station by pressing Transfer, then the BFS key. If you're a secretary and you want any
calls to the boss's phone to come to your extension instead,
just press the BFS key on your phone. It sets up his phone to
forward all calls to you. You and anyone else who has a BFS
key to it can see by the flashing BFS key that his calls are
being forwarded. The frustrating part
comes in when you try to copy a set that has a BFS key,
relocate one of the sets using Automatic Set Relocation, or
move the set. It just won't work, and you get an error. Secretaries often don't fully understand
the BFS key a day or two after being instructed (maybe that's
my fault). They invariably get calls forwarded and can't
figure out why. I still like the key and recommend it for
certain situations, but I expect to have a few calls that
indicate training is necessary again. Q: When I'm talking
on my 2616 set, I sometimes want to mute the handset and
listen while I talk with someone else in the room. I know I
can put the call on the speaker and mute it, but I want to
keep it on the handset. How do I do that?
A: You don't. At least, you don't with
a 2616 set. If you have release 25 software, you can look at
moving to the new 3904 set. It has that capability. Q: I have seven
phones set up with the same extension as SCR (Single Call
Ringing), but I'm trying to change them to MCR (Multiple Call
Ringing). When I try to change one, I get an error.
A: You can't have a combination of SCR
and MCR for the same extension, even while changing them. You
must NUL (null) all of the instances of the SCR before
programming the MCR. Before you do
that, though, are you sure you want to use MCR? In North
America the feature has a bug that doesn't appear in the
international versions of the software. The effect is that
only one call can be ringing at once on the MCR keys. The
second caller hears a busy. If this is a sensitive application
where you never want to get a busy signal, you may want to
find another solution. Q: When I look at the FFCs (Flexible
Feature Codes) in LD 57, they're all blank. I want to know the
default codes for features such as Call Park and Call Pickup,
but they don't print. How can I tell what the default codes
are?
A: I couldn't find a good list
in the NTPs, so I got this list from another user. I'm giving
you the LD 57 abbreviation, the code, and the feature name. CDRC: *46, CDR Charge Account Code CFWA: *21, Call Forward All Calls
Activate CFWD: #21, Call Forward All
Calls Deactivate CPAC: #76, Call Park
Access Code CPRK: *76, Call Park C6DS: *30, Six-Party Conference ELKA: *57, Electronic Lock Activate ELKD: #57, Electronic Lock Deactivate HOLD: #40, Permanent Hold CFHO: *60*, Call Forward/Hunt Override RPAX: *81, *82, Radio Paging Access RPAN: #82, Radio Paging Answer PUDN: *43, Pick Up DN PUGR: *44, Pick Up Group PURN: *42, Pick Up Ringing Number RCFA: *22, Remote Call Forward Activate RCFD: #22, Remote Call Forward
Deactivate RDLN: *80, Redial Last
Number RDNE: *53, Redial Number
Erase RDSN: #54, Redial Saved Number RDST: *54, Redial Store RGAA: *37, Ring Again Activate RGAD: #37, Ring Again Deactivate SPCC: *51, Speed Call Controller SPCE: #50, Speed Call Erase SPCU: *50, Speed Call User SSPU: **, System Speed Call User TFAS: *83, Trunk Answer From Any
Station LILO: *40, Login-Logout for
500/2500 ACD Sets NRDY: *45, Not Ready
Activate/Deactivate for 500/2500 ACD Sets GHTA: *48, Group Hunt Termination
Allowed GHTD: #48, Group Hunt
Termination Denied
Voice
Mail
Q: My Meridian Mail system must be haunted. When I listen to messages
or leave someone a message using my speakerphone, it interrupts me or
suddenly decides to "send" the message or other actions. If I pick up the
handset, it seems to work ok. What can I do?
A: For starters, you could just use the handset when you're in Meridian Mail.
No, that was too easy. Your Meridan Mail is probably hearing other sounds
in the room and interpreting them to be tones, as if you are dialing commands
from the phone. It may be a radio, or overhead paging sounds, or any other
noisemakers that you may not even notice. I had a problem like that with another
brand of voicemail some years ago that interpreted the voice of one user
as tones. When I explained to her what was happening, she altered the
pitch of her voice talking to voicemail and the problem went away.
Q: One executive here says his Meridian Mail mailbox is haunted.
Messages just disappear before he gets to hear them. I'm not getting
such complaints from anyone else in the company.
A: There are two answers to this question. The quick and easy one is
to have the manager change his password and DON'T TELL ANYONE the new
password. Really. Chances are someone else is checking his or her
messages. This may be someone who guessed the password (or saw it dialed),
or it may be a secretary or family member.
The other solution is to be used in cases where you want detailed information
about the voicemail settings. Log into your Meridian Mail using the password
TOOLS instead of your regular one. It will prompt you for your password
afterward. Then select Session Trace. You will be able to step through
all activity on the mailbox for a set amount of time. Read through all the
information carefully, and you can tell when each activity occurred, what
happened, and in many cases what extension or telephone number placed the call.
It's almost as good as listening in on the sessions.
Q: I need help building a menu in Meridian Mail.
A: Here's one I like. It allows
callers to log into their mailboxes, leave messages by number
or name, and dial extensions by number or name. I have to make
some assumptions, so you may need to adjust several things to
make it fit your situation. Assume you
have extensions in the 4xxx range. Use 4444 for this menu. In Meridian Mail, start in 3-Voice
Administration, 4-Voice Services Administration, and
4-Thru-Dial Administrations. Select Add, use Thru-Dial ID
0001, Title is 4xxx Extensions. Dial By Number, DN Length is
Fixed 4 Digits, Left Pad 4, Suppress Prompt No. The
Restriction/Permission Set should be one that only allows
internal calls. Add a second Thru-Dial
Definition with an ID of 0002 and a Name of Names. Dial by
Name, and Restriction/Permission Set again that only allows
internal calls. Next to 3-Voice
Administration, 4-Voice Services Administration, 6-Voice Menu
Definitions. Select Add, use Voice Menu ID 0003, Title is
General Menu. Skip down to Key 1, Action VM, Comments Login
Mailbox. Key 2, EM, No Mailbox ID, Comments Leave Message. Key
3, Action TS, Thru-Dial ID 0002, Comments Dial By Name. Key 4,
Action TS, Thru-Dial ID 0001, Comments Dial 4 . Set all the
unused keys to RP. Move your cursor to the No. that follows
Greeting Recorded (Voice) and select Voice. Enter the
extension number of a nearby phone, press Enter, and answer
the phone. Select Record, and use a greeting like this:
"Welcome to Sprockets Inc. You can dial your extension at any
time. Press 1 to log into your mailbox. Press 2 to leave a
message for someone. Select 3 to dial by name." Select Stop,
and review your recording. Be sure to save it when you get
through. Next to 3-Voice
Administration, 4-Voice Services Administration, 1-Voice
Services-DN Table. Select Add, use Access DN 4444, Service MS,
Voice Menu ID 0003. In the PBX, LD 23,
REQ NEW, TYPE ACD, ACDN 4444, MWC NO, MAXP 1. Skip down (keep
pressing Enter) to NCFW, and enter the main number for your
voicemail. That should do it. When you
dial 4444, the menu plays. Test all the choices before giving
the number to anyone. Make sure callers can't reach an outside
line through BARS or access codes. You
can add a toll-free number that points to 4444, if you like.
You can also add hidden functions that are not given in the
greeting. Q: I'm
trying to build new mailboxes in Meridian Mail. I've added a
couple of new Classes of Service and they show up in the
Administration screens, but when I go to build a new mailbox
the only option it gives me for Class of Service is Personal.
Is there something I have to do to activate the new Classes?
I've already shut down and re-started the mail, but it didn't
seem to do anything useful.
A: Yes.
You need to tell the system which Classes of Service to offer
when a mailbox is programmed. From the Main Menu, select
General Administration, then General Options. Add the new
Class of Service numbers in the line that says Class of
Service Selection. Q: I recently changed the primary DN (key
0) of a 2616 set. There is no voicemail box associated with
the DN, nor the old DN that was on that particular phone, yet
the light is lit. I reset the phone and also searched for any
phones where the primary DN differs from the MWI DN and found
nothing. How can I get the message light to go off?
A: There are two issues here: turning
off the light and keeping it off. To turn off the light, do
this. Change your personal voice mailbox so that the message
light DN (Directory Number) is the number now programmed as
key 0 on the phone. Call your number and leave a message for
yourself. Now call voice mail and retrieve the message, and
any other messages you have. The light should go off. That
takes care of the immediate problem. Next, you need to check to see whether any
other mailboxes are set to turn that light on and off. You
said there are no mailboxes with the current Prime DN, so
(assuming you have Meridian Mail) use the Find feature to
determine whether any of the mailboxes has the Prime DN listed
as the Message DN for that mailbox, and if you find one,
change it. That should take care of the problem for awhile. SystemsQ: Our new Meridian
1 system will be different from the old Rolm system everyone
is used to. Can I set up the system to work like the Rolm so
the transition can be easier?
A:
There are a couple of areas where you may be able to ease the
transition for users. One of them depends on the voicemail
system you have. If you elected to use Call Pilot, you may
choose a set of feature codes that mimic the Rolm PhoneMail
operation. If you're using Meridian Mail, you won't have that
flexibility. The operation of the phone
system will never be exactly the same as the Rolm system, but
you can modify the feature codes your users will access to
simulate their familiar environment. For instance, you can
make the Group Pickup code **3 and the Directed Call Pickup
code *3, just like the Rolm systems I used to work on. You define the Flexible Feature Code block
in LD 57. Check your documentation to find the features that
will relate to the practices of your users. TrainingQ: Is there any
reason to use Nortel's training instead of another company,
like Global Knowledge?
A: As a
matter of fact, Nortel recently outsourced its training to
Global Knowledge. And yes, there is an advantage to using
Global Knowledge. If you need Nortel certification, that's
where you get it. If authentic certification is not necessary,
other companies provide adequate training for some
products. Miscellaneous
Q: If I connect my Core card to Ethernet, can I FTP to the PBX to do
administration?
A: No. There is no FTP daemon running in the PBX. Nortel wants you to
buy Meridian Administration Terminal or its successor, Optivity Telephony
Manager, for this.
If you connect the PBX to your Ethernet LAN, be sure to block broadcast
messages from reaching the PBX by using a router. A router seems to be the only
acceptable way to block those nasty broadcast messages that can take your
PBX completely out of service.
Q: My dial-up
connection to the PBX works fine, except for the rude
interruptions that occur on the hour. I'll be typing in
changes, and at the stroke of the hour, line after line of
numbers spew onto my screen. It keeps me from the work I need
to do. What can be done about this?
A: It sounds as if you are receiving
traffic reports through your modem connection. If you don't
have a device calling into your modem to receive these
reports, you can turn them off for the modem port. First, you'll need to know which TTY port
is used for the modem. In later software, the system will
indicate which port you've reached when you first log into the
system. That information in hand, you can print the
Configuration Data Block in LD 22, at Gate Opener ADAN. Look
at the TTY for the modem and see which entries follow the USER
prompt. TRF indicates Traffic will be sent to this port. To change it, use LD 17 and Gate Opener
ADAN. After the prompt USER, enter XTRF and whatever other
entries you need to remove irritating messages you don't
need. Be aware, though, of what
messages you turn off. These may come back to haunt you, so be
prepared to add them back. Q: We play classical music for music on
hold. We're considering using the radio instead, but people at
the radio station said we might need to pay licenses to use
it. Why doesn't the radio station want us to use their
material?
A: You will be
disappointed with this answer. You probably are already
broadcasting copyrighted music without a license. The owners
of the performance on your classical recordings have probably
listed them with either BMI or ASCAP. You need to either get a
license to use the material, or begin using material that
doesn't require a license. Contact them at www.bmi.com or
www.ascap.com to get more information. I want to say just a bit about this
license issue. For music performers or others who receive
royalties on music, this is a sensitive issue. You have chosen
their music to play day after day for your customers. This
implies that you receive some benefit from the playing of
someone else's music for your business. You should really feel
obligated to pay the royalties due to these creators of the
music you provide for your customers who wait on hold. If you
don't, then you should feel obligated to avoid the big fines
your company will pay if you get caught. Q: Can I have more
than one kind of music on hold in my Option 61? People don't
seem to agree on the music style.
A: I don't think I can help you pacify
all the listeners, since few situations allow the user to
listen to his or her choice of music on hold. You could only
offer that choice when calls are under the control of CCR or
Symposium, by adding the menu item to your script. I have personally arrived at a choice that
gets few complaints. I play orchestral CDs of John Williams
conducting his movie themes and also Disney songs. I avoid any
singing, since this gets many complaints (it seems everyone
hates at least one style of vocal music). I also eliminated
one CD that was otherwise very good, but had an extended drum
solo on one of the cuts. You can choose
a different music source for each trunk route. This is easy,
since the route programming includes a prompt for the music
route. Just set up additional music sources that terminate
into trunks in new music routes, and use those route numbers
for different trunk routes. You build
new music routes in LD 16. Print out your current route in LD
21 and duplicate it when you create a new one. You'll need a
trunk port for each, set up like the one assigned to your
current route. You will also need a CD player or some kind of
music source to connect to the trunk you built. Q: Surely there's a
better way to handle the administration of my Nortel Option
11C. My vendor installed this dumb terminal with a printer
attached, but scrolling back is difficult and very limited.
With some trouble I can print out information on a dot matrix
printer, but it needs some Windows features!
A: I agree. The terminals don't
provide a state-of-the-art interface. I have a PC connected to
a serial port on mine, and use Procomm Plus to talk to it with
a VT 320 emulation. I have the session defined for a HUGE
scroll-back buffer, and the mouse makes it much better. I can
scroll back and view days of activity at a time. With your Option 11C, you can create a
backup of the system on your hard drive through this PC, and
you can connect the PC to your network to use network
printers. You can also get MAT
(Meridian Administration Terminal) for a Windows-based tool.
If you use Optivity for your data equipment monitoring and
administration, you'll be glad to know Nortel will soon
release Optivity Telephony Manager, the successor to MAT. Q: Occasionally, I
set up a 500 set to forward to an outside number so we can
dial the extension and reach the outside number. So far, the
only way I know to make it work is to program the 500 set,
then clip a test set onto the port and dial the codes to
forward calls to the outside number. Isn't there a way to do
this using the administration terminal?
A: Yes, there is a much better way.
Create a Phantom Loop (if you don't already have one) and
program a Phantom TN for the application. Program a Default
CFW for the set. The line will look like this: FTR DCFW 11 918008795866 In the above example, the 11 tells how
many digits, and the following number is the one that will be
dialed. If you have Remote Call Forwarding set up, you can
even change the destination number remotely. Q: Where can I
download upgrades to the firmware in my Meridian HomeOffice II
routers and line cards?
A: I don't
know. Nortel moved the files, and I haven't found anyone who
knows where they are now. If any readers can help me locate
these, please send me an email.
Phil has worked
on both sides of the house, having been a field technician
servicing Nortel and other PBX and key systems, and also
having managed networks of PBXs. He is active in his local
SL-1 Users organization and has attended INNMUG. Phil can be
reached via email at philruffin@hotmail.com.
Reprinted with permission from Telecom
Reseller magazine.
Definity Demystified
by Walt Medak
Trunk
Card
Q: Our facility has grown outside our campus and we must expand to
another location several blocks away. We have been told that we can
connect our current Definity to an Expansion Node if we can run
fiber-optic cable between the two sites. Our initial concerns of having
the remote die if there were a major problem with the main node was put
aside with the proposal of an EPN w/Survivable Remote. Are there any
other options to connecting the two sites besides having to pay for the
installation of a quarter-mile of fiber?
A: An Expansion Processor Node (EPN) is one way to connect two
physical locations to have a single virtual office. Calls can be made between
sites via extension-to-extension, one Attendant Operator can field calls
for both locations, etc. A better alternative would be to install a separate
Definity G3(x) at the remote site and connect them via T-1 utilizing the
DCS features of the Definity. There might be more cost involved for software,
but my guess is that it would be close to a wash against an EPN with
Survivable Remote. It is a survivable remote in itself, and much, much
easier to administer on-going and has all the benefits a Survivable Remote
EPN can give.
Q:
We constantly are being called by numbers of companies selling
long-distance and local alternative service to our current
providers. We are unsure of what to believe, and the sales
people we have been talking to don't ever seem to be
knowledgeable about the Definity system. Is there any rule of
thumb for how many and what kind of trunks we need?
A: That's a big question with multiple
answers, but mostly a fairly simple one. The Long-Distance
"Fair Wars" have been going on since the '70s, and have had
various incarnations as to what's the best/cheapest mode of
trunking. We've gone from strictly CO trunks to WATS trunks,
to Tie trunks, and on and on. For most systems in this day and
age, one trunk group would suffice for Local and Long-Distance
calls using a switched "Pick". Many providers are offering
that L.D. service for about ten-cents per minute, making the
installation of any other trunking, such as a dedicated L.D.
T-1, more costly than it's worth unless your call volume is
sufficiently large enough to substantiate that overhead.
That's where most of the Long Distance "Agents" who are
calling you are weak in expertise, and you should talk to a
consultant or very knowledgeable tech. Use the "list
measurements" command to determine your current trunking
usage, and your call-accounting package, if you have one, to
determine if you are over or under trunked. The BIG thing to
be aware of is to combine all the trunk groups you have into
as few as you can, such as one or two. Doing this will require
fewer trunks in your system. Q: I am installing a new ISDN-PRI two-way
DID trunk group in my Definity-G3i, and am not able to get the
trunks to work. The D-channel status is "in-service", and the
provider says they show their D-Channel and trunks as
in-service, and that we must have a problem with our switch.
The Definity shows the trunks as "OOS-FE PINS". What are we
doing wrong?
A: Probably nothing. I
have this "thing" about network providers pontificating their
absence of responsibility during events such as this. The key
definition of where the problem lies is in the "FE" portion of
OOS-FE PINS. FE stands for Far-End, and that may be where the
problem lies, or in your not defining the protocol they should
be using to match the Definity. Essentially, the Definity best
uses Protocol Version "a", which is located on the DS1 form
under the Country Protocol. "a" is known as "custom", and "b"
is known as National (or NI-2). If you have defined the
Definity as "a" (the default) and the provider is defined as
NI-2, you will get exactly the results you are getting. In
some cases, the provider isn't sure of what "custom" is, and
will fight to get you to change the Definity to NI-2. I have
had problems with this, and have always reverted to "custom",
which they can usually support. If you get resistance to this,
get one of our staff on line to work with the provider. They
can normally convince them to see it our way. Q: We are moving
our system to our new corporate offices, and are changing our
trunking in the process. Currently we just have Central Office
analog trunks and a T-1 from our Long Distance provider.
DID(Direct-Inward-Dial) trunks over a 2-way T-1 by our network
provider was proposed to us, and though we think we would like
DID, are unsure of what the heck it is that's being proposed.
Will a 2-way DID trunk group work on a Definity, and what do
we need to do to administer it?
A:
DID is a protocol that most network providers and PBX
manufacturers have agreed upon and conform to for delivering
calls directly to an extension without intervention from an
Operator or Attendant. It works very well on the Definity, and
even the System75. It's a protocol dating back as far as
processor driven telephone systems. Essentially, the call is
proceeded with the last 3 or 4 digits of the telephone number
from the network provider to the Definity, which then
identifies them and routes them to the extension, hunt-group,
etc. that has those matching digits at its number. That's one
of the nice things about the design of the Definity; it uses
extension numbers as its means of addressing. You have nothing
to administer once the trunk group has been assigned, as it's
seamless in the Definity. The 2-way part is a function of the
T-1 and the network provider's switch allowing both incoming
and outgoing calls to utilize any trunk for either. It is a
very good service that usually has a substantial cost savings,
and I would highly recommend it. This is a very brief
description of the protocol, and if you are still confused,
give me a call so I can better explain it in detail. Attendant
ConsoleQ: Our Attendant Console is not able to
take any more than one call at a time. It used to be that we
could put one call on hold, and the next call waiting would
come in on the next loop button. Is there some option that has
changed? We can't find any that seem to make a difference.
A: We have just recently been exposed
to that very problem. It was the 48-volt power to the console
that did the trick for us. Originally, the console was powered
from the switchroom via the white-brown pair of the station
wire. We disconnected that, and installed the optional 48-volt
transformer with a 400-B adapter for the power source, and the
problem went away. Q: We experienced a power failure, and now
our Attendant Console isn't able to forward calls for our
president and sales manager as it did previously. We have
looked at the software assignments for "Attendant 1" and it
all looks the same as the printout we have of it when it was
installed. What else can we look for?
A: You probably won't find anything
wrong with the software for "Attendant 1", or anywhere else.
Power failures can cause strange problems, some which you will
never recover from without replacing costly hardware. In your
case, however, I would bet you will recover by simply
powering-down your system, waiting a minute or so, and
powering it back up. Don't forget to shut down your Definity
Audix first, if you have one. Auto
Attendant
Q: How do those folks with an auto-attendant that says
"to dial by name, press 1" do that with an Intuity?
A: First, create a second-level auto-attendant. On the
second page of that auto-attendant, in the upper left corner,
is a field asking which addressing option, either "extension
number" or "names". Choose "names". Then go to the third page
and type an "e" in each of the destination fields for "1" through
"9" and make their transfer type "transfer", meaning to send this
call back to the PBX. You will need to record the greeting for the
new auto-attendant as something similar to, "Please enter the first
three letters of the person's first name, followed by the # sign" (if the
names are entered in the mailboxes as "John Doe"; otherwise if they are
entered as "Doe, John" record the greeting to say....."the first three
letters of the person's last name followed by the # sign"). You then
send one of the choices of the main auto-attendant to that second-level
auto-attendant's extension number by utilizing the "call-answer" rather
than "transfer" method. The "call-answer" routing keeps the call in the
Intuity rather than transferring it back to the PBX which would then
have to transfer it back to the Intuity causing an undue delay in the
whole process.
Q: Our Auto-Attendant has several layers
to it, and we seem to have a long delay between selections. I
have listened to other Auto-Attendants, and they don't seem to
have this problem. What are we doing wrong, or what do we need
to correct this?
A: Assuming your
Auto-Attendant is on your voice mail that's in the Audix
family, it seems that on the 3rd page of the Auto-Attendant
the "Treatment" for your selections might be set to "T" or
"Transfer" instead of "CA" or "Call-Answer". If you are using
the "T" or "Transfer" option, you need to change it, as what
is happening is that you are sending all choices back to the
PBX for further transfer back to Audix each time. Changing the
option to "CA" or Call-Answer" will send it directly to the
next Auto-Attendant box with little or no delay. Q: We re-record our
Audix auto-attendant two or three times a year, and have a
need for doing it at this time. The person who always did it
is no longer with the company, and we have no idea where to
begin. How do we find what is needed to re-record?
A: The recording of an auto-attendant
is nothing more than recording a voice mailbox in the Audix;
it is done exactly the same way. If you know the extension
number of the auto-attendant, you have all that's necessary to
do the job, assuming you know how to change the 3rd or
"selection" page of the attendant. To some, this page is
abundantly clear as to its function, but to others, due to a
lack of familiarity, it may not be. If you have a problem you
will need to either consult your documentation, which is not
usually an easy feat, or consult your service company or
myself at the addresses below. If you don't know the extension
number, give the command "list attendant" at the Audix command
line and a list of all the auto-attendants will be displayed.
If the originator of this auto-attendant was doing the job,
the name of it will be readily apparent. If not, and you have
a number of auto-attendants from which you cannot distinguish,
you will have to start at the PBX by looking for the extension
number that sends it to the Audix, the method of which,
fortuitously, is answered by the next two questions in this
column. Q: Our
network provider is changing our service from analog trunks to
a T-1, and we are adding DID numbers at the same time. They
have asked us what numbers we currently have answered by our
auto-attendant that we may need as additional DID numbers to
avoid losing those calls. How do we find out all of the
numbers we have answered by our Intuity auto-attendant?
A: Calls come into a Definity PBX in
only one way: through a trunk group. By mentioning you are
adding DID numbers at this time, I assume you have no analog
DID trunks currently, so therefore most probably have only
Central Office (C.O.) trunks at this time. If so, that makes
things easy, as on the first page of each trunk group, on the
right-hand side of the page near the top, are the "incoming
destination" and "night destination" fields where the
information you are looking for is contained. You will also
need to look at the port assignment page (either the third,
fourth or fifth page, depending on your release of software)
to get the list of telephone numbers assigned to this trunk
group (again, assuming the person who set this trunk group up
did their job and listed them properly), and to see if the
"night" field has an entry. If it does, this is another
destination that may be pertinent to your network provider. If the telephone numbers aren't on the
port assignment page, you will need to have the ports traced
to determine the numbers in that trunk group. What your
network provider is most probably looking for are any numbers
your callers use to contact you, known as LDNs (Listed
Directory Number) that you may have either advertised or given
to your callers in mailings, business cards, etc. To continue
to get those calls, you will need to have them included in
your DID string or list, as they are associated with one or
more of the analog trunks as the trunk's telephone number.
When those trunks are removed, that number will be lost if you
don't have it changed to one of your DID numbers. You will
then need to build some entity, most likely an "x-port"
station covering to the destination previously found in the
trunk-group "incoming-destination" field, or better yet, if
you have vectoring, utilizing that. There are many variables
here, so if I have missed answering your question, please
contact me. Least Cost
RoutingQ: Our Least Cost Routing, or ARS as
Lucent terms it, seems to have some bugs in it. We make calls
that return fast-busy signals or recordings that utilize the
same route patterns as calls that go out successfully. Are we
omitting something?
A: It sounds as
if you are making the proper comparisons to successful calls,
so I'll assume you know how to utilize the ARS fairly well.
The fast busy's could be coming from your system or the public
network. If you are certain they are coming from the public
network, and never have problems other than with the area code
or prefix you are trying, your problem is most likely with the
programming in your local or long-distance providers' systems.
I have never experienced a properly programmed ARS having
problems other than on the public network, and feel that there
are no bugs in your system. SystemsQ: We have just entered our new budget year and are looking
to upgrade our System 75 R1V3. The budget isn't so great that we
can do this under the proposal we received for the Definity G3V9.
Is there any way to upgrade in a gradual manner and not have to jump
right to the latest release? Our other sites have Northern Telecom
and we are able to do that with those systems.
A: Bless your heart for asking that question.....that's what the
secondary market is all about! In order to upgrade through the OEM
you would probably have to jump to the current release, which is the
V9, but through the secondary market there are values to be had for
earlier releases of the Definity. I personally like either the V4 or
V6. The V4 is an exceptional value, but has the downside that it is the
last release to use the older TN786B processor. Although the V6 uses a newer
processor, it is only going to be usable up through the V8, as the V9 has
an altogether different processor. Since I'm betting the primary reason
you wish to upgrade is driven by budget availability rather than additional
feature needs, my suggestion would be to upgrade to the V4 because if you
want to later go to the latest and greatest you will have to replace either the
V4 or V6 processor anyway. You can expect to save well over 50% on the system,
and as far as the software license goes, Lucent/Avaya hasn't defined that
very well other than what one client has related to me: "If you want Lucent/Avaya
support, it must be re-licensed." Especially on things like this, if you
need more information, we'll be more than happy to explain the reasons why the
secondary market is great.
Q: Our Definity
G3sV3 maximum capacity has been reached. Our answer for
additional capacity has only been to upgrade. I have talked to
other company's system administrators who tell me they have
systems earlier than mine with three times the stations we
have. What is different about our system that requires an
upgrade to get the larger capacity?
A: It's the "s" in G3sV3 that
indicates that your Definity has a designation of "small". The
good thing is that you have, I believe, a system that was
installed before the licensing of port sizing came into
effect, and thus can probably expand your system to the G3iV3
(full capacity) by simply adding one circuit pack. It's the
memory that defines the size of a system, and with the
addition of the memory circuit pack CPP1, your system will
magically become a G3iV3. The CPP1 is a daughter board that
attaches to the TN786-B Processor circuit pack. If you are not
sure of any of this, you should contact a knowledgeable
interconnect company, or it may be available from Lucent. Or,
you may contact me at any time with your questions. Q: We are a
self-maintained facility that has its own technicians and
systems analysts for our Definity G3iV4. We used to be able to
perform the tasks of testing, busying-out and releasing, etc.
Since we have discontinued our maintenance agreement with
Lucent, we have lost those permissions. How do other
self-maintained Definity users cope with this?
A: This has been a thorn in the side
of many, many Definity end users for over a decade. We have
heard through the years many stories of what it took to get
those permissions back from Lucent. In the end, it's Lucent
you will have to deal with, but we have known companies that
paid a one-time fee for activation of MSP (Maintenance Service
Permissions), and others who claimed they were charged monthly
for the privilege of doing their own maintenance. There are
also different levels of MSP - one for stations, one for
trunks, and one for processor/common-control. I have witnessed
end users with combinations of some or all of them, and have
heard from some that they were not able to get all of them. My
observation has been that the larger the company, and
logically, the larger the technical staff, the more likely it
is to be granted the processor/common-control MSP. This will
have to be taken up with Lucent, however, and it seems the
permissions and answers vary from region to region of the
country, and is possibly dependent on size, or perhaps
tenacity, of your company. Q: I can no longer call forward my calls.
Instead of confirmation tone, I get another dial tone part way
through the process. What has gone wrong with my system?
A: Probably nothing. Usually this is
caused by having "Console Permissions" added to your Class of
Service. Check your COS and see if you have this service. If
so, you will need to dial the call forwarding feature access
code, then the extension number you want to forward, and then
the number to which you want to forward. This feature allows
an individual to forward or cancel forwarding for extensions
other than just themselves. It is usually reserved for System
Administrators. Q: We have a phone that's exposed to the
public on which we need to be able to call an outside pager,
but don't want it to be used for any other calls outside of
our system. Is this something our Definity can do?
A: Yes, that's a task the Definity can
easily perform. Here is a little-used but nice way of
restricting phone calls. First, you need to establish a COR
that has "All-Toll" as its Calling-Party-Restriction. That
will bring up a field about half-way down the left column
called "Unrestricted Call List". The number entered here will
need to be defined as the one allowed by the Toll-List. It's
entered much the same as the ARS entries. This is something
that will be obvious when you look at it, or will leave you as
mystified as you were before, depending on your experience
level with the Definity system. There are many steps needed to
successfully accomplish the restriction, so if it isn't clear
how it's done feel free to call me or one of our staff at the
number listed below. Q: Is there any information stored in the
Definity-G3i to identify calls to our system in cases of
security or maliciousness?
A: There
are many things that can identify these types of calls. First,
if they are ongoing, report them to your network provider and
ask for assistance in identifying the callers. For decades,
the phone company has been able to "trap" malicious calls if
trunks were configured for it. Getting them configured may be
the difficult part, but be persistent. You may not be provided
the information, but if you report the problem to the local
police they can obtain the information from the network
provider and prosecute if you are willing to sign for it. Other means internally is known as either
SMDR (Station Message Detail Recording) on early versions of
your system, or CDR (Call Detail Recording) on later ones.
Both SMDR and CDR need to be set up in software before
collection of calls begins. You will find these under "change
system features" on the 2nd or 3rd page for SMDR or under
"change system cdr" for CDR. You will also need to make sure
your trunk groups all have SMDR or CDR optioned to "y". To get all of this straight, you will need
to consult your "Upgrade & Additions" manual (or give us a
call at the number below). Then you will either need to
connect your system to a serial printer, a PC using a
communications application allowing a "capture" mode, or a PC
utilizing a call-accounting application. Even with all or part
of this, you will need ISDN-PRI trunks to identify where the
call originates, and many of these perpetrators know how to
block their identification. For short-term use, if you don't
already use a call-accounting package, I would recommend using
a PC and a communications program such as ProComm in the
capture mode. It's quick, easy and least costly. Q: We are opening
small offices of 15 to 30 people in locations across the
country. Many of them will be connected to our regional
offices which all have Definity-G3's. For these smaller
offices we are considering either Partner, Merlin Magic, or
Norstar key systems. What will be needed to connect any of
these systems to our Definity network?
A: Another big question with multiple
answers probably too long for this column. For starters,
virtually any modern phone system can connect to any other
modern phone system, with differing ease. It might be that you
need to dial a different code for each location, and then the
extension number, or it could be as easy as dialing the
extension number and letting the Uniform Dial Plan software
route the call for you. Normally, staying within the same
manufacturer’s product line will garner better support both
during installation/implementation, and ongoing. I would like to suggest a system you
didn't mention, a System75 R1V3 or Definity-G3V4 (or lower)
from the secondary market. With any of these systems you will
be close to the same costs, but will also be able to take
advantage of the unique networking capabilities, not to
mention the important significance of like architecture,
allowing for movement of your hardware from location to
location as size, closures, relocations, etc. occur. Q: We're an
interconnect very familiar with the Definity system, but are
having trouble making DCS/UDP work between two sites. We think
we have programming in place for all that the manual shows,
but still can't call from one to the other. Any quick
suggestions?
A: There's nothing
quick about implementing DCS/UDP, and to even attempt it shows
you must have good familiarity with the Definity. And as you
also know, there's no way I will be able to address every
possible problem you could have, as there are dozens of
entries required to implement them. However, the most common
problem I have noticed is the need for the trunk groups and
their trunk-access codes to be the same in both switches not
being adhered to. I would bet this is your problem. Let me
know, please. Q:
We have just upgraded our Definity-G3 from a V3 to a V8. Our
vendor, an independent telephone company not associated with
Lucent, says it can no longer support us because they cannot
log in to the system except with our "cust" login, which has
no maintenance permissions. Is there any way we can continue
to have our vendor log in with maintenance permissions?
A: In an earlier edition of this
column I hinted at this problem. Lucent has added security
levels to the V6, V7 and V8 releases of software that are very
good at keeping anyone but them out of the system at levels
needed for system maintenance. The security method is a great
thing if you want to make sure nobody but Lucent is able to
get into your system. If, however, your interest is having
others than Lucent maintain your system you have only one
option - that is to pay Lucent to add maintenance permissions
to your "cust" login. This can be quite costly, and,
coincidentally or not, would probably make your maintenance
costs rise beyond what Lucent would charge you if you just
used them. Did your upgrade include anything you needed beyond
the capabilities of your V3, or was it just too good a deal to
pass up? If it were the latter, it's another reminder that
there's no such thing as a free lunch. Q: We are a small
company that has no dedicated system administrator. Recently,
due to some customer complaints, we found that two of our
trunks had been disconnected for several months. We noticed
the Definity identified these disconnected trunks, but didn't
light our alarm lights on our attendant console. Is there any
way of the system notifying us when this occurs?
A: Yes, there is a way to make
trunking problems identify as "minor" alarms, thus lighting
the minor alarm light on the attendant console. You would need
to have your vendor enter into the "set options" screen so
that trunking minor alarms are reported as such, instead of
the default setting. This downgrades trunking off-board alarms
to "warning" alarms, which alert nobody. Q: We are about to
add another Definity system, which we intend to network to our
current one. It is a secondary market purchase, so we are
attempting the software implementation ourselves. We are also
going to have to change our dial-plan from 3-digit to 4-digit
dialing in order to utilize DCS and UDP. Is there an easy way
to accomplish this without hand-entering each extension
number, and if not, are there any pitfalls we need to
avoid?
A: You are attempting a
courageous mission, but one that's easy to perform with some
time, patience and with paying detailed attention to the
following steps. First, there is no fast and easy way (other
than letting us do it for you), but it's not difficult either.
I would suggest to first print out all existing stations,
data-modules, hunt-groups, terminating-extension-groups, and
vector-directory-numbers (i.e., "display station 200 count
1000 print"), and for quick reference, a "list station print".
You will lose the personal-abbreviated-dialing lists of the
stations, and if you want to re-enter them for the 4-digit
stations, you will need to print them out, too. Then,
beginning with the first extension, give the command "list
groups-of-extension (extension number)", and note what groups
the extension belongs to, as you will have to put the new
4-digit extension back in that group (this is probably the
most forgotten step, and causes more problems than all the
rest combined). Then "duplicate station (station number)" and
add the new 4-digit station number with a port assignment of
"x" and the same name as the 3-digit station. Next do a
"remove station (3-digit station number) and note the port
number before pressing "enter" to remove the station. Then
"change station (new 4-digit station number)" and add the port
that was on the 3-digit station in place of the "x" you
originally entered. It is at this point you must go to the
groups the station belonged to and enter the 4-digit station,
if any, and don't forget the personal-abbreviated-dialing
lists if you are going to replace them. I would be happy to
discuss this with you if it's too convoluted here; just
call. Q: We have
a System75 that was removed from an office we closed some time
ago, and now want to install it in another location. Our
account rep has told us we will need to upgrade to a
Definity-G3V8 because there is no North American Numbering
Plan in our ARS. Is this the only thing we can do with our
System75?
A: Not by my standards. I
felt the NANP upgrades were about 50% useless at the time
everyone was scrambling to get upgraded before the axe fell.
First, remember ARS lost the usefulness for its intended
purpose when long-distance network providers gained the PIC
capabilities. It does have a good means of restricting
calling, but there is another method that was intended for
that very purpose, so ARS isn't really necessary for that.
There was a time when it was normal procedure to dial a 9 for
local and an 8 for long-distance. If you institute that
criteria (only needed if you have long-distance provided over
a different trunk group than the local trunk group), then you
have no need of ARS whatsoever if you give the local trunk
group a TAC of "9" and the long-distance trunk group a TAC of
"8". Remove it. Get rid of it. Kill it. 'Nuf said? Now use
your System75 for at least the next five or more years and
spend your telecommunications budget on necessary things. Q: We constantly
hear about the TCP/IP capabilities of the Definity G3v8 and
wonder what we are losing if we don't upgrade to it. What's
your opinion?
A: I view it about
the same as video teleconferencing. Did you install one? If
you did, you're probably among the 70% who hardly ever use it,
and maybe among the many who have disconnected it. If you
didn't, have you missed it? Probably not. The attribute of
TCP/IP is its functionality over your LAN. That means you can
send your voice calls over the same wiring that your PC
network uses. If you are a mega-corporation with a new
installation of a campus somewhere, that's probably an option
to consider. If you are a mid-to-large corporation, even if
you are contemplating moving to a new location, do you know
how much wiring you can install for the cost of an upgrade and
the hardware necessary for your Definity to become LAN
compatible? I think TCP/IP is going to be a thing for all end
users to consider one day, but I don't think that day has
arrived yet for the masses. It's another of those bleeding
edge commodities at this time that's probably best left to
those who have bulging budgets and a curiosity that dictates
living on that edge. Someday, however, I bet even I will join
'em. Hunt
GroupQ:
Our hunt group application is angering many of our field
personnel who call into it, and I need a better solution.
Currently we have each agent in our office assigned as the
only member in his own hunt group so we can put the call in
queue if he is active on a call. After several minutes we have
the call transferred to another hunt group comprised of all
ten of the agents. At times this may take a caller out of
queue and place him at the tail end of another, which makes
him unhappy. How can I keep the queue positioning and yet have
the call escape to be answered by another agent if one is
available?
A: This is a common
problem with small call-centers. Because you are small doesn't
mean you don't need sophisticated solutions. This is always a
dilemma, as it usually equates to dollars. What you need is
called Expert Agent Selection (EAS), utilizing Call Vectoring.
This method of call-center processing is capable of look-ahead
routing which determines who is available for the call and
sends it based on criteria you can define, including the
agent's ability to manage the call. It also allows the caller
to escape to a voice-mail box or elsewhere if they don't want
to remain in queue. EAS and Vectoring, along with an
Announcement circuit pack is the answer to your problem, but
application of it would take up this whole column, so I
suggest you contact your vendor, or call us for further
information about it. The downside of it is that Lucent's
licensing can be very expensive. Voice
MailQ:
How does one maintain two extension numbers but only one Audix
MailBox? We are a police agency and have occasions to be
called on more than one number, but would like the convenience
of maintaining only one MailBox for multiple members of our
force.
A: I hope I'm not going to
show my ignorance here, but I really could only come up with
one solution. That was to have one of the numbers forward to
voice mail in normal fashion, and have the other cover to a
remote coverage point (i.e., r1) that was sent outside the
system by dialing 9 and then the DID number of the first
extension. One can also try covering to a hunt group that's
been remote-call-forwarded (by the attendant or user with a
COS that has "operator permissions") to the first extension.
Good question..... any readers have answers to this one? Q: Is there any way
we can forward voice mail messages between our Definity Audix
system at our corporate office and our DuVoice system located
on our Definity at another site?
A:
Both of those systems support the AMIS protocol, which was
designed specifically for what you are trying to do. AMIS
(Audio Messaging Interchange Specification) basically places a
call from one voice mail system to the other, accesses the
proper mailbox, and the transferring system plays the message
while the other system records it via a plain old telephone
call. It seems to have a bad rap as an archaic method, but
those systems we have implemented were very successful,
especially in light of the nominal cost compared to digitally
networking two Audix systems. AMIS is an underused and mostly
unknown method of transferring messages between systems. Based
on our luck with it, we highly recommend trying it. Q: Our Audix Small
8.2 is serving us well, but we have been told by Lucent/Avaya
that we will not be supported any longer. What is the
secondary market doing for its support, if anything?
A: Happily, upon the notification you
mention, the secondary market is awash with hardware and
knowledge to support the Audix Small (or Large, for that
matter) in any software release for years to come. I say
"happily", for the hundreds of secondary market Distributors
and Dealers are delighted it is being abandoned by
Lucent/Avaya. That system is, as it has been for years,
functioning quite well, and serving the needs of the end users
needing only voice-mail and/or automated attendant. From here
on out, the upgrading to more storage and ports will only get
more economical, as many who were scared into upgrading to
Intuity Audix have traded-in their old systems, and
Lucent/Avaya has auctioned them to the secondary market
distributors, and this has caused the pricing to drop
significantly. It's a great place to stay for quite awhile if
you don't need any other options than voice-mail or
auto-attendant, or need to be on the "bleeding edge". All that
is necessary is to scan the advertisements in Telecom Reseller
to find a secondary market dealer who will, if not themselves,
refer you to a dealer who can support you as well, if not
better, than you currently enjoy (enjoy?.........
whatever!). PhonesQ: We have need for
wireless telephones in our manufacturing plant. We have tried
the Transtalk9000 series phones a few years back, and they had
too great a failure rate and were noisy with interference on
them. Are any of the cordless phones on the market compatible
with the Definity?
A: Yes, just
about all of the cordless phones, including the new
2.4Gigahertz models, are compatible with analog ports on the
Definity. However, the new Lucent Transtalk 9031 model is less
problematic than the older ones you tried and seems less
susceptible to interference. In addition, there is the
wireless module that emulates cellular capability in the
Definity and allows wiring these antenna "sites" over just
about any campus or industrial complex of any size. Q: We have two
Attendant (Operator) positions that share incoming call
responsibilities. At times such as breaks there will only be
one position in operation, and the other on "Position Busy".
We have an 800-Number that comes in on a hunt group where both
of the Attendants are members. However, when either position
is on "Position Busy", it still receives incoming calls for
the 800-Number. The other numbers in the trunk groups don't
have this problem. What can we do to correct this?
A: There are two methods to correct
this. The problem stems from the fact that the "Position Busy"
button has no effect on anything but the console queue, and
since you are using a hunt group, there is no way of knowing
the position is indeed "Busy". You can either get rid of the
hunt group and put its extension number in the
"Listed-Directory-Number" screen or you can remove the
attendants from being members of the hunt group, give the hunt
group a coverage path that covers ALL calls in the coverage
criteria, and then make point one of the coverage path "0" or
"attd". This will then send these calls to the attendant queue
which recognizes the "Position Busy" status of each
attendant. Q: We
have calls going to a conference room phone for which we can't
trace the origin. How do we find out what calls are directed
to an extension?
A: The first thing
to determine is if it is a DID number. If so, that's probably
your problem. Make it a non-DID number and no outside calls
should be able to reach it. The second place to look is to see
if it's a member of some group that gets calls. You can do
that by giving the command "list groups (extension no.)". The next place is to see if it's listed as
a destination of a trunk group as mentioned in the answer to
the question above. It might also be a coverage point for some
other station, which should show up in the "list groups"
query. You can also "list coverage-path" to see if it shows up
as a destination. And lastly, and the most difficult to find,
is to see if it's a point of a "route-to" in a vector, which
would require looking at all the vectors one-by-one. On the
Far-Side, it's also possible it's the night-destination of a
hunt-group, and your system is in the night mode when calls go
to it. Some systems are left in the night mode all the time.
If you can't find the problem with these hints, give me a call
and tell me my answer in this column flunked, and I'll help
you find it. MiscellaneousQ: Are there any alternatives in the secondary market for certified
Definity training? We are looking for the possibility of on-site training
for both system administration and end user training, or at least an
alternative to traveling to Denver each time we want to train an administrator.
A: I'm not sure of all of the "certified" folks that are available,
but there are many who have been trainers for Avaya/Lucent who are no
longer employed there who on their own offer training of both types you
are looking for. They are located in most major metropolitan areas of the
United States, though are not usually of high profile and are therefore not easy
to contact. I hereby extend an invitation for those folks who read this column
that are trainers, or those who may know of a trainer or training company,
to contact me and I will list their names in a future issue. There are also
companies in the secondary market who offer training, as we do, but like us,
only in our facility at this time. We hope to suitcase the class for major
metropolitan seminars in the future, but that's no help to you now.
Q: As a technician
for an Interconnect, I am frustrated with the endless finger
pointing when either trying to repair or install a T-1.
Arguing is nonexistent without test results from a T-Bird or
equivalent test device, whose price leaves them out of the
question for small interconnects. Is there any way of testing
within the Definity to prove where the circuit problem
lies?
A: Yes, the Definity has a
terrific method for positive identification, but
unfortunately, it's buried inside the Tier-II's login, and not
below that. It allows for a set-up, watches the call and
reports where and what happened to it. Most of us have no
access to that login, so the next best method, and one which
has never failed me in testing hundreds of T-1's, is setting
the "Synchronization" screen's primary source to the T-1 you
are testing. Then, in succession, give the commands "disable
sync", "set sync <DS1 Board location>, and finally
"enable sync". Once the last command is completed, do a
"status sync". If the T-1 has connectivity, it will show
connected to the DS1 board, otherwise it will show connected
to the Tone-Board. If you show Tone-board synchronization,
check all the physical connections onsite, i.e., the CSU
connections, PBX cable connection, etc. If all of your
connections look good, replace the DS1 or the cables or the
CSU or all of them, and test again. If you have the same
results, your problem will undoubtedly be with the network
provider. This has never failed me. You may think having all
of this spare equipment is extreme, but if you don't have it
with you, you have no business attempting the job, and it's
more cost efficient than purchasing a T-Bird. Remember, this
is for the T-1 only, so if you are having trouble with the
trunking using the T-1 facilities, that will be another
problem entirely assuming you are synchronized with the DS1
board. Q: We
have three sites with Definitys, all connected with T-1's.
These sites also have our WAN connected via their own T-1's. I
have heard that we could combine these facilities to reduce
the number of T-1's to accomplish this. How is this done?
A: In the United States, a T-1 is made
up of 24-channels, each 64K in bandwidth. Your question didn't
specify how much bandwidth your WAN required. If you can do
with 64K bandwidth between each site, the easiest way to do
this is to remove one of the voice channels from the voice
T-1, and add a tie-trunk group with that channel as the only
member, and its destination a data extension whose port is
connected to a 7400C data module. The Definity PBX is a very,
very good data switch. If, however, you need more bandwidth
than 64K, you will instead need a multiplexer that will split
out the channels you specify to an RJ48 interface for the
Definity, and a V.35 interface for your WAN. The cost for
these units is approximately $2000.00. Q: We have just
been informed by our long-distance provider that we are
getting a large number of calls from a foreign country, some
with very long call durations. They have told us to check to
see if we are being used as a switching center for these
calls, sending them out of our system to various points in
this country. Where do we begin?
A:
There are many ways these clever people have discovered to
gain entry to PBX systems and back out again to make
long-distance calls at your expense. The most obvious
protections deny the transfer of calls that come into a system
to go back out without at least human intervention. Convenient
means have been programmed into your system to allow you to be
able to transfer yourselves through, but it leaves you
vulnerable to thieves, just like leaving valuables in your
unlocked car. You need to make sure you lock your system, and
don't leave any keys where hackers can find them easily. First of all determine whether your system
needs any automated capability to transfer incoming calls back
out. If not, eliminate trunk-to-trunk transfer in the system
parameters features screen. If you do need some activity in
and back out, utilize COS and COR restricted trunk groups
accessed via Remote Access that is controlled with restrictive
COR's and long "barrier codes", and utilized "account codes".
Also, much of the toll fraud is accomplished via Voice Mail
systems. In the AUDIX make sure you have "enhanced call
transfer" selected or set to "yes". Most of all, have your
system reviewed by a consultant who specializes in toll fraud.
Lucent offers a great toll-fraud protection program, though
I'm not sure if it's only for their maintenance clients. Q: What is the
function key setup for administering a Definity remotely with
a PC?
A: The easiest method is to
answer "4410" at the "terminal type" line where it shows a
default of "513". This is where you login to the system and it
asks you for the terminal type. This method will assign your
function keys for you upon each command. If you want to use
the "513' terminal type, the function setup is as follows: (I
forget some, but the essential ones are these, all case
sensitive) F-1 = escape+Ow for "Cancel", F-3 = escape+SB for
"Enter", F-5 = escape+Om for "Help", F-7 = escape+[U for "Next
Page" and F-8 = escape+[V for "Previous Page" (the O's are
capital o's, not zero, and the [ is the left bracket). The
"escape" is different for different programs, but many, such
as Procomm Plus, use ^[ (upper-case 6 and a left-bracket) for
the escape sequence. For example, this would make "Next Page"
program as ^[[U. This, too, can be confusing, so a call to us
might help clear any cobwebs I may have spun. The very best program we have found for
administering either a Definity or an Audix system is
"Definity Site Administration" available from Lucent/Avaya.
DSA, as it is known, has the function keys assigned for you,
whether you use "4410" or "513" emulation. Until this program
came out, we advised against using anything but Crosstalk
Mark-IV(out of production) or Procomm Plus. Terranova was a
confusing mess, though I understand recent versions of it work
OK. Q: We have
been proposed an upgrade that will enable us to use TCP/IP
functionality with our Definity. This has been defined to us
as the best option for distribution for our PBX, and touted as
a necessity. What exactly is the advantage for us?
A: It's hard to define, not knowing
your size and condition of flux. If you are a large (several
hundred or a few thousand port system that is installing in a
location where new wiring is necessary, there may be an
advantage to it. If you are located where you have all the
wiring in place for your system, it doesn't make much sense to
me to upgrade just to have TCP/IP. My understanding is that
you can use the TCP/IP functionality to send your voice
communications over your data wiring in prioritized packets
eliminating the need for double wiring; i.e., a run for voice
and a run for data to each workstation. If my information is
correct , TCP/IP only works well on a LAN. Assuming it will
work well some day over a WAN, then there may be good reason
then to upgrade to it to reduce the cost of networking both
voice and data systems separately, assuming the prioritized
packets don't slow your data network to unusable
functionality. Based on the information above, my
recommendation would be to wait for functionality over a WAN
before considering it necessary, unless it will save
cost-justifiable dollars in additional wiring to the LAN. If
any of our readers can add information supporting or contrary
to the above, I invite them to contact me, and I will run the
information in the next issue. Q: We received the following question from
Dawn Paolino:
We have reached the
capacity of bridged appearances on our 8434DX phones, or so we
have been told. We understand there is a maximum of 24 lines
(main number and rollover) including 1 CAM. We now have to add
two phones to an assistant’s desk to handle coverage. Does
Lucent have another product available that will allow the
assistant to know who is being called so they can answer,
"John Doe's office; how can I help you?"
A: Yes, there is another way of
identifying covered calls, and I believe it works much better.
However, it does require the use of display phones at the
covering location. I am a strong anti-bridged-appearance
implementer. If there is any way of doing things without the
use of bridged-appearances, I believe it to be a better way. I
never use them! I'm as passionate about that belief as a vote
counter in Florida. If you use a voice terminal at the
covering location that has a display, the covering reason is
given on the display in easy-to-understand code. First, it
will say something like "Local to John Doe" or "Dawn Paolino
to John Doe", and then in the far-right side, there will be a
code of either a "b", "d" or "s". The "b" means the called
voice terminal is busy on another call, and the call is
temporarily bridged at that called voice terminal. The "d"
means the call was not answered at the callers terminal in the
prescribed number of rings, and that the call is temporarily
bridged at that called voice terminal. The "s" means all calls
at the called voice terminal are covering because their
"send-calls" button is pushed. In as much as you are using
8434DX voice terminals, you already have the capability, and
the information should already be appearing on your displays
if only you would use call-coverage instead of those
!@#$%^&* bridged appearances. Q: From Arron Meyer, some information
about TCP/IP was received:
I work for
an Avaya Business Partner and wish to comment on your TCP/IP
information in the November 2000 Definity Demystified
column.
The TCP/IP functionality in the
G3 is used mainly for integration with adjuncts at this time.
When R9 becomes available, Avaya's 4600 series of IP
telephones also becomes available for use. The reason that a
switch upgrade to the G3 is usually installed by us, in my
experience, is simple: the customer is adding a new adjunct
that requires IP integration, such as CenterVu CMS or Intuity
Audix. Why? Mainly because X.25 integration is being phased
out. The 7400D data module will no longer be manufactured
after December, effectively ending a key product for X.25
Processor Interface. Avaya will no longer support new
installations of adjuncts using X.25 switch integration. It
will continue to support customers who already have adjuncts
integrated using X.25. Also, if the customer is a G3csi
(Prologix), the only option for digitally integrating an
adjunct such as Intuity Audix is TCP/IP using a
TN799/TN799B/TN799C Control-LAN board. The only other option
is Mode-Code integration as the G3csi platform does not
provide for a Processor Interface circuit pack. For IP phones
or IP trunks, at MedPro (Media Processing) board will also be
required.
A: Thanks, Arron for your
response, but from it I see further reason that TCP/IP's time
has not yet come. Definity R9 is now available, as are its
4600 series of voice terminals, but they still only allow
TCP/IP connectivity over a client's LAN, so the only advantage
I can see is the elimination of dual runs of station wiring,
which if already in place (which was the case in the question
in the column) makes it a moot point. And the view of most of
us who appear in this forum is that the OEM isn't the only
place a client can find the products they need, as the
secondary market is awash with perfectly good equipment with
the same or better warranties and guarantees. The idea that
products such as the 7400A, B, C or D data modules are
effectively ended because of the OEM's decision to quit
manufacturing them just doesn't hold water. And I love the
fact that Avaya will no longer support new installations of
adjuncts using X.25 integration, as there are hundreds of us
out here who can and will. Avaya's a good choice, but not the
only one. As for the Prologix, it's a
product whose time never did come. For less than the cost of a
new Prologix, a client can obtain a secondary market G3s with
all the same or better coverage. And Mode-Code integration
works flawlessly, as many third-party manufacturers of
voice-mail products will agree. On the issue of IP trunks, or
MedPro boards, you are ahead of me on those, and I will have
to defer to you on them, though I think perhaps its relevance
to most clients will be negligible as a reason to upgrade to a
software version just to include TCP/IP. Thanks for the
information Arron, and let's keep the dialogue going to better
inform Definity Users and Administrators, as I found your
information very educational.
Walt Medak is
president of Medak & Associates, Inc. He can be contacted
by phone at 800-452-6477 X5001 or by email at walt@medak.com .
Reprinted with permission from Telecom
Reseller magazine
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