 |
Avaya IP Office Installation & Programming Support |
Q. |
I have installed Avaya IP Office Manager on a PC that is directly connected
to a LAN port on my IP Office, but I still cannot communicate with the IP Office. What do I do? |
A. |
The Avaya IP Office’s
default IP address is 192.168.42.1, and out-of-the-box,
it only knows how to communicate with other devices in
the same address range (192.168.42.x). Your PC probably
has an IP address assigned outside this range. Program your
PC to have a static IP address of 192.168.42.2 and try again. |
| |
|
Q. |
Do
I have to reboot the Avaya IP Office phone system every time I make changes?
What programming changes require that I merge the config and when do I have to reboot? |
A. |
There is a
long list of mergeable and non-mergeable changes. A good
rule of thumb is this: if you have ADDED something (ie. an
extension, a hunt group, a new user…) you’ll
have to reboot. If you’ve just CHANGED something
(ie. button programming for a user, changed a short code,
changed an incoming call route…) you can do a merge
instead. You will automatically be prompted to merge or reboot in later releases of the IP Office software. |
| |
|
Q. |
My Avaya IP Office software
licenses keep coming up as “invalid” in the IP Office Manager program.
How do I make them valid? |
A. |
Make sure
all of the following are in place in order to get your
licenses to come up valid:
- Installation of Feature Key Dongle - Your feature key dongle must be securely connected to
the PC that acts as your license server (often also the
VoiceMail Pro server)
- You have installed the Feature Key Server software
from the IP Office Admin CD on the PC with the dongle.
- The Key Server service is running on the key server
PC (sometimes stopping and restarting this service
helps).
- The correct IP address for the PC acting as the
Key Server has been entered in Manager under System->License
Server IP address.
Avaya IP Office Phone System Licenses are unique to a specific dongle. If you have
multiple dongles, make sure you have the right licenses
installed in Manager for the dongle you’re using.
|
Q. |
I am using Vonage ("Vonage"), a service provided by Vonage Holding Corporation (NYSE: VG), for our dial tone provider and Caller ID signaling rarely works. Why? |
A. |
Our lab test results show that Vonage Caller ID is not 100% compatible with the Avaya IP Office phone systems. Packet8 service, provided by 8x8 Inc. (NasdaqCM: EGHT), has proved to be a much better alternative to Vonage for voice over IP dial tone, as their Caller ID works so much more reliably when interfacing with the Avaya IP Office system in that it provides not only the callers' name and number almost 100% of the time. On the other hand, Vonage tended to provide acceptable caller ID information, ie. callers' phone number and name, once every ten calls at best.
| |
| |
|
 |
Avaya IP Office VoiceMail Pro Support |
Q. |
I’ve
installed Voicemail Pro and pointed the IP Office’s
voice mail IP address to the server where I installed it,
but voice mail still isn’t working. Why? |
A. |
Check the
following:
- Make sure you have completed installation of the VoiceMail Pro software and the VoiceMail Pro Service is running
on your voice mail server PC. Sometimes stopping and restarting
the service will help.
- Make sure that your Voicemail Pro license is coming
up as valid in Manager
- Make sure your voice mail server IP address is correctly programmed
in Manager (set in systemàvoicemail tabàvoicemail
server IP address. Make sure voice mail type is set
to "PC" in this tab as well.)
|
Q. |
VoiceMail
Pro is working, but the GUI takes a very long time to
load. I also don’t have “telephony handset”
as an option for recording prompts in the GUI. What’s
up? |
A. |
You’re
probably using an early version of the IP Office Manager that does not support Windows 2003 Server with VoiceMail Pro.
Make sure you have upgraded to the latest Avaya IP Office software release. |
| |
|
Q. |
My
voice mail messages cut in and out, and so does the “voice mail prompt
lady” when she’s talking. How do I get rid
of this? |
A. |
Voice mail
jitter can come from one of two things: an overloaded
server and collisions between packets on your network.
Check the following:
- If you are using a 403 base module, your network interface
card in your Voice Mail Pro server PC needs to be set to 100/Half Duplex.
This setting should also be used for any ports
on 3rd party routers that the server and/or the IP
Office are using as an uplink to the rest of your
network.
- The voice mail server should be dedicated to voice mail
and not running any other services. Do NOT install
VoiceMail Pro on a server that is also running Exchange,
hosting websites, or any other high-impact services.
- Make sure all screen savers, power save and hibernation modes are turned off.
-
- Anti-virus software can often impact the VoiceMail
Pro service as well. Disable any background anti-virus
software on the voice mail server PC.
|
| |
| |
|
 |
Avaya Phone Manager Pro & User Utilities |
Q. |
I’ve
installed Phone Manager, but when I run it, it won’t
let me do anything. |
A. |
You need
to configure Phone Manager to log into the IP Office phone system
to control your phone. The first time you run Phone
Manager, do the following:
- Go to configureàPBX
- Enter the IP address of your IP Office in the PBX
address field.
- When you enter the correct IP address, the Username
box should automatically fill up with the names of
every user in the system. If it does not, you have
an incorrect IP address or your computer cannot communicate
over the network with the IP Office.
- Select your name from the list and enter your user’s
password where indicated. When you click ok, Phone
Manager should come up, as “you” and all
the buttons and menus will work.
|
Q. |
Phone
Manager logs in, but it only comes up in Lite mode. |
A. |
Make sure
that:
- You have valid licenses for Phone Manager Pro
- “Phone Manager type” is set to “Pro”
under your user’s telephony settings in Manager.
- You don’t have more people trying to log
in as Pro users than you have licenses for. If you
have 5 Pro licenses, the 6th user to log in will get
Phone Manager Lite, even if the above settings are
correct.
|
Q. |
I
can’t use the “call number back” feature
in Phone Manager because I have to dial a 9 to get an
outside line. Can this be fixed? |
A. |
Yes. However,
the setting is not in Phone Manager. Open your phone
system config in IP Office Manager, go to your lines area, and
select a line. The “prefix” setting is designed
to fix just this problem. Whatever you enter in the
prefix area will pre pre-pended to the caller ID of
every call that comes in on this line. Now the entire
caller IDs in your logs will have 9’s in front
of them and you can call them back using the “call
number back” feature. |
| |
|
Q. |
Ok,
that works for the 9, but what about putting a 1 before
long distance calls? |
A. |
A short code can be programmed to support this functionality. Contact Mountain States Telecom for more information about this programming technique.
|
| |
|
Q. |
I
was told I could dial my phone from Outlook and other
3rd party programs. How can I get that to work? |
A. |
There are
just a few simple steps to start dialing from Outlook:
- Make sure you have installed the TAPI drivers on
your PC from the User CD.
- Make sure the TAPI driver is configured correctly.
You can find the settings by going to the Windows
control panelàphone and modem optionsàadvancedàAvaya
IP Office TAPI2 service provider. (if you don’t
see that last part, you have not installed the TAPI
drivers).
- In the properties of the TAPI driver, make sure
that “single user” is selected, and that
the switch IP address, username, and password are
identical to the settings you use under the configureàPBX
window of phone manager.
- If you had to change settings in the TAPI driver,
you must reboot your computer for them to take effect.
Once the driver is configured correctly, you have to
tell a program from which you wish to dial to use the
driver instead of your PCs modem. In Outlook, here’s
how you do it:
- Open a contact you wish to dial. They must have
at least one phone number field filled out.
- Click the auto-dialer button in the toolbar of
the contact.
- Click the “dialing options” button
- Select “IP Office phone:<extn #>”
under the “connect using line” drop-down
box.
- When you click “start call” your phone
will automatically dial the contact number. Note that
all rules about dialing 9 and/or 1 still apply.
|
| |
| |
|
 |
Avaya Voice over IP Telephones |
| |
To
get started, please refer to the 4600 or 5600 Voice over IP telephone installation
guides and the 4600 LAN administrator’s guide to
resolve a large portion of networking issues related to
the Avaya 4602, 4610, 4620, 5602, 5610 and 5620 Voice over IP telephone sets. |
| |
|
Q. |
Do
I need a Voice Compression Module (VCM) to have Voice over IP telephones?
I’ve been told that I do, and that I don’t,
and that sometimes I do and sometimes I don’t…help! |
A. |
We’ll
make this easy - YES. Your Avaya IP Office system MUST have voice
compression hardware in place in order to do any sort
of VoIP telephony (IP phones and/or tying multiple IP
Office units together.) All of the Avaya Small Office Edition
control units come with VCM hardware built-in. On the Avaya
403, 406, and 412 models, you MUST purchase an additional
VCM card in order to do Voice over IP functionality.
The confusion is arising around WHEN the system uses
the VCM and when it does not. Depending on how you are
going to use the system, we may be able to reduce the
SIZE of your VCM, but not eliminate it all together.
One of Mountain States Telecom’s knowledgeable
engineers will be glad to help you determine how big
a VCM you’ll need. |
| |
|
Q. |
Everything
is working with my phone, but I keep getting a “TFTP
timeout error” when it boots up. Why does it do
that? |
A. |
The phone
looks for 2 files on the TFTP server every time it boots
up: 46xxsettings.scr and 46xxupgrade.scr. By default,
the upgrade script exists and the settings file does
not. The phone doesn’t NEED the settings file
to work, but it will give you a timeout error if it
can’t find it anyway. To resolve this:
- On your TFTP server create a new file in c:\program
files\Avaya\ip office\manager called 46xxsettings.scr.
- You can leave this file blank, or you can put in
a line, starting with a “#” symbol, stating
that the file is only there to get rid of timeout
errors on the phones.
|
Q. |
I’m
trying to get my phone to work on a VPN. When I plug in
the phone it just says “TFTP ping” and never
goes any further. What’s wrong? |
A. |
Your phone
can’t communicate with the other end of the VPN
tunnel – usually because of an incorrect “router”
setting. Make sure that the IP address you are giving
the phone for its router is the address of whatever
device is handling your LOCAL end of the VPN tunnel. |
| |
|
Q. |
I’m
trying to get my phone to work on a VPN. My phone talks
to the TFTP server, and the phone prompts me for an extension,
but after I enter the extension it just says, “discovering”
and never goes any further. What’s wrong? |
A. |
Your phone
can talk to the IP Office, but the IP office can’t
talk back. This is due to incorrect or missing IP routes
in the IP Office configuration. The actual IP routes
you need to add will vary based upon your network topology,
but a good rule of thumb is that you should ALWAYS have
a default route:
- IP address = blank
- Mask = blank
- Gateway = address of router through which the IP
Office can communicate with other networks (often
this is the device handling the IP Office end of the
VPN tunnel)
- Destination = LAN1
|
Q. |
When
I call another user with a Voice over IP telephone, I can hear
them but they can’t hear me. How come? |
A. |
Chances
are you are talking on two phones with a router between
them. To fix the one-way talk issue:
- Open your phone system config in Manager
- Go to the extensions area
- Select the extensions in question. On each, go
to the VoIP tab and make sure that “enable direct
media path” is NOT checked.
|
Q. |
I
cannot do a group page to my Voice over IP telephones. My digital telephones
do group pages just fine. Why? |
A. |
This is
one of the limitations of Voice over IP telephones. In the current
state of the technology, group page simply does not
work at this time. |
| |
|
Q. |
I
have an Avaya IP Office – Small Office Edition. I’m
trying to get VoIP phones to work on LAN2 and cannot.
How do I make this work? |
A. |
Please call Mountain States Telecom at 719.635.0006 for support on the Small Office Edition. |
| |
|
Q. |
I’m
using Manager to act as my TFTP server for my IP telephones.
It used to work, but now when the phones boot up I just
get a “boot – unable to process” message
in the TFTP log in Manager. I haven’t changed anything
since it was working. What’s wrong? |
A. |
We’ve
seen this happen, too, and we’re not sure why.
For this reason, Mountain States Telecom highly recommends
that you download the free Avaya TFTP Suite Pro application
and use THAT as your TFTP server application instead
of leaving a Manager window open all the time. |
| |
| |
|
 |
Avaya 3616 & 3626 Wireless IP Telephones |
Q. |
What’s
the deal with this Voice Priority Server (SVP)? I am only using
a few IP wireless phones and I can’t see voice quality
being an issue. Do I HAVE to have an SVP? |
A. |
Yes. While
its true that you could probably get by without doing
QoS on most small networks with just a few phones and
minimal data traffic, the truth of the matter is that
the 3616 and 3626 look for the SVP as part of their boot
sequence. If they don’t see an SVP on the network
they will give you a “No SVP found” error
and die. The phones WILL NOT BOOT without an SVP visible
on the network, period. |
| |
| |
|
 |
Overhead
Paging Systems |
Q. |
I’ve
downloaded the job aid for attaching 3rd party paging
equipment, I’ve followed it exactly, but all I get
is a busy tone when I try to make a page call. How do
I make this thing work? |
A. |
Paging systems
are tricky. You will almost always need some sort of
an interface unit to go between the IP Office and your
paging amplifier. The best solution is to purchase a
Universal Paging Access Module (UPAM) from Avaya. However,
there is one trick you can try:
- Connect the POT extension on the IP Office you
wish to use to a simple telephone cord splitter (available
at Radio Shack, etc)
- Connect one of the outputs of the splitter to the
input on your paging amplifier.
- Connect the other end of the splitter to a cheap,
single-line analog telephone. Leave this phone OFF
HOOK ALL THE TIME.
|
| |
| |
|
 |
Other
Miscellaneous Problems |
Q. |
If
I dial the phone too slow, the call doesn’t go
through. I either get a “your call cannot be completed
as dialed” message, or the call just drops all together.
Can anything be done about this? |
A. |
Yes. This
is a common complaint when using a PRI. Open your telephone
system configuration file, in Manager, go to the ‘system’
area, click the ‘telephony’ tab. These two
values will be of interest to you:
- Dial Delay Time: This is the amount of time that
the system will wait after the last key you press
before attempting to interpret what you dialed. Increasing
this value will make the system give you more time
to dial a phone number.
- Dial Delay Count: This is the number of key presses
the system will wait for before trying to interpret
what you’ve dialed.
The optimal settings for these two values will depend
on your environment (how your T1 works, whether you
have to dial 10 digits on all outside calls or not)
so there is no “correct” setting for all
cases. With some simple trial-and-error, however, you
will be able to use these two settings to optimize your
dialing.
|
| |
|
Q. |
I
have some users who keep getting logged out of the groups
they belong to. I keep enabling their hunt group membership
in Manager, but they keep getting logged out again. What’s
going on? |
A. |
There are
several places that people can log in and out of a group,
and they will conflict with each other if set differently.
You can enable/disable a user’s membership in
a hunt group in any one of the following places:
- The “agent mode” tab in Phone Manager
Pro.
- The properties window for a hunt group in Manager.
- Via a programmed soft key on the phone.
- Through the menu on the 44xx or 46xx series phones.
(Avaya 4412, 4424 system telephones, the
5410, 5420 digital telephones and 5610 and 5620 IP telephones.)
That last item is often the problem and is the last
place most people look. Press the menu button on your
44xx, 46xx, 54xx or 56xx-series phone and use the arrow keys to
scroll over until you see “group” on the
screen. Make sure there is a “ ^ “ character
over the word “group.” If not, press the
button under “group” to make the character
appear. Otherwise, the phone will constantly be trying
to log out of whatever groups it’s user belongs
to. |
| |
| |
|
 |
General Voice over IP Telephony ( VoIP ) Support Questions |
Q. |
What Bandwidth Do I Require for Each Voice Call? |
A. |
The bandwidth used varies depending on the compression method chosen. Avaya IP Office supports a wide range of compression standards, including the most popular G.723.1 and G.729a. These will occupy approximately 10K and 13K of bandwidth respectively. |
Q. |
What Delay is Acceptable? |
A. |
Try to keep the overall end-to-end delay to 150 milliseconds or below. An idea of the delay inherent in the network can be measured by carrying out a ping test and dividing the result by two. IP Office has built in echo cancellation to maximize speech quality. |
Q. |
How Do I Minimize Voice over IP Garble and Clipping? |
A. |
Garble, clipping and some distortion quality problems are symptoms of variable delay and or packet loss. Variability in the delays of traffic is called jitter. Jitter and packet loss may be the result of switches and routers that are either faulty or working outside their design intentions.
Avaya IP Office provides jitter buffers to compensate for a moderate amount of jitter found in networks. Voice traffic is quite tolerant of small amounts of packet loss so in most cases this may be ignored. Where packet loss is excessive (greater than 2% say) the cause should be established and fixed. This could be due to a fault or simply an over worked device discarding packets. Significant packet loss can cause perceptible losses in speech, to the extent that no speech may be heard either in one or both directions. |
Q. |
How Do I Minimize Distortion? |
A. |
Each time speech is converted into a digital signal and back again, tiny difference from the original creeps in. The more times this happens on a single call, the bigger those differences can become. These differences can become perceptible as distortion. Ideally, the path speech takes should only require one 'analog to digital to analog' conversion and this will be the case in many instances. Exceptions to this occur when making calls to mobile telephones or voice mail systems where the analog to digital to analog conversion may occur twice (once on IP Office and once on the mobile network, etc).
Different encoding methods will have different effects. Avaya IP Office supports a range of encoding methods to allow you to choose the one with the right quality versus bandwidth for your network. In general multiple conversions should be minimized wherever possible. |
Q. |
How Do I Minimize Delay Induced Echo? |
A. |
Delay in a network originates from a number of different sources and phenomena. A primary source of delay is the process of converting speech to VoIP traffic. The Avaya IP Office supports a number of standards based encoding methods to allow the optimum trade off between quality and bandwidth to be made. IP Office incorporates integral echo cancellation to minimize the effect of echo introduced in the Voice over IP conversion process.
Another source of delay comes from data and voice traffic queuing at the ports of switches, routers, gateways and or bridges that make up the network. It is possible that the traffic queuing at a port is minimal and no action needs to be taken. This would be the case if the available bandwidth far exceeded the demand. To overcome queuing bottlenecks in the network, IP Office prioritizes voice traffic using a standard known as DiffServ. This marks each IP packet carrying voice with a flag so that routers, etc. can force packets containing voice to the front of the transmission queue. An alternative method of prioritization that can be used by switches and routers, with an equally satisfactory result, is to look at what protocol is being used and prioritize this. All voice traffic is carried using two easily identifiable protocols, RTP and RTCP. Both methods are equally good, choose whichever method is the most cost effective and easiest to implement and manage.
A similar source of delay can be attributed to specific network nodes that convert from one network medium to another. For example T1 trunk lines may be carried across a high speed DSL like connection and converting from the high speed link back to T1 in the access gateway takes time to perform. Any Voice over IP ( VoIP ) traffic being carried through this link is therefore subject to the delay introduced by this conversion step. The delay may be minimized by ensuring that an appropriate QoS mechanism is enabled in the gateway to prioritize the Voice over IP traffic. IP Office incorporates integral echo cancellation to help minimize the effect of this kind of delay introduced by the network.
Delay can also be introduced as a consequent of collisions occurring on particular segments of the LAN. Collisions result when two devices on a shared switch port or segment try to transmit simultaneously. This causes all devices to stop transmitting for a period of time. This is a normal characteristic of many older Ethernet networks and, if occasional, may pass unnoticed. The more devices sharing a switch port, and the busier they are, the greater the opportunity for collisions. This is simply resolved by reducing the number of devices on each port, or by dedicating a port to each Voice over IP device. If you are just using Voice over IP telephony to link two IP Offices together, it's well worth dedicating a port to each IP Office and router at either end of the link as the cost implications are likely to be minimal. In this regard it is important to dimension a network to cope with existing traffic demands as well as any future increases in traffic carrying capability.
|
| |
|