Frequently Asked Questions |
Click here to see sample device configurations     
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What is VOIP
(Voice Over Internet Protocol)? |
Voice over
Internet Protocol (VoIP) is a technology for communicating using
“Internet protocol” instead of traditional analog systems. Some VoIP
services need only a regular phone connection, while others allow
you to make telephone calls using an Internet connection instead.
Some VoIP services may allow you only to call other people using the
same service, but others may allow you to call any telephone number
- including local, long distance, wireless, and international
numbers.
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How VoIP Works
VoIP converts the voice signal from your telephone into a digital
signal that can travel over the Internet. If you are calling a
regular telephone number, the signal is then converted back at the
other end. Depending on the type of VoIP service, you can make a
VoIP call from a computer, a special VoIP phone, or a traditional
phone with or without an adapter. In addition, new wireless "hot
spots" in public locations such as airports, parks, and cafes allow
you to connect to the Internet, and may enable you to use VoIP
service wirelessly. If your VoIP service provider assigns you a
regular telephone number, then you can receive calls from regular
telephones that don’t need special equipment, and most likely you’ll
be able to dial just as you always have.
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How do
I contact IP Communications customer support? |
Sales and Billing Questions
North American Customers: Call 1.800.588.2350 option 1
International Customers: 1.678.460.1475 option 1
Customers may also send emails to:
sales@ipcomms.net .
Technical Support Questions
North American Customers: Call 1.800.228.8596 option 2
International Customers: 1.678.460.1475 option 2
Customers may also send emails to:
noc@ipcomms.net
or open a ticket online at
http://www.ipcomms.net/support
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How do I get started? |
Getting started with IP Communications is easy. Simply visit
our web site at
www.ipcomms.net and choose
the service and package you desire. Simply choose the phone
numbers or number locations you require and complete the order
process. Within 24 hours (usually less) of receiving your
order, you will receive a service provisioning letter that will
contain all the information you need to configure your VoIP device.
Configure your device with our information and begin placing and/or
receiving calls. You will be able to monitor and maintain your
service online using AMI our Account Management Interface found
here:
www.myipcomms.net.
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How do I buy a phone number? |
For
most packages, you can add a phone number to your account at any
time. Just log into to AMI (the online account management
interface
www.myipcomms.net ) and follow the
links to add numbers to your account.
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Can I try the service
"Risk Free"? |
You can try us for 30 Days "Risk Free".
If you are dissatisfied with you IP Communications service for any
reason within the first 30 days of sign-up, you can cancel your
service and we will give you a full refund. See full
details
Try us for 30-days "Risk-Free
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What kind of call quality does IP Communications offer? |
Although
our network is used by both businesses and individuals, it has been
designed to meet the requirements of large corporations as well as
local and long distance carriers. The end result of quality depends
greatly on the network you are using, but in general you should
expect to receive quality equal to if not greater than that of a
traditional phone service.
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How long
does it take to activate my new service?
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In most
cases new service can be activated and ready for use in 24 hours or
less. Unique or larger orders may take more time.
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How long does it take to add numbers or ports to my existing service? |
In most
cases service additions can be activated and ready for use in 24
hours or less.
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Billing/Payment Questions |
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How do I pay my bill online? |
Simply enter a
valid credit card or use your Pay pal account. You will be charged
for your initial order. Your card will also be automatically charged
once a month if you select a monthly plan. If you purchase a monthly
plan, you can cancel at any time, just fill out a service
cancellation form and return it to
sales@ipcomms.net. We
require 30 days notice to cancel your account
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How do I pay using Wire Transfer (bank
to bank transfers)? |
A wire transfer
is a transfer of money from one bank account to another. The actual
transfer is done by the bank, and neither the sender nor the
recipient of the money sees or touches the actual funds. Here are a
few steps to transfer money from one account to another.
Contact your bank by phone or via the Internet and provide the
following information
Wire Info: IP Communications, LLC.
Bank Name: Bank of America / Route No:
061000052
Act No: 003279334736 / SWIFT: BOFAUS3N
2597 George Busbee Parkway,
Kennesaw, GA 30144
Email:
sales@ipcomms.net
Reference*: "Reference should be your IP Communications Account
number"
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How do I pay using Company Check? |
Please remember IP Communications is a
pre-paid service, we do accept company check, but you will be
responsible for insuring that you leave enough time for mailing
complications or unforeseen issues. To mail your payment
simply mail to :
Mail To: IP Communications, LLC.
1925 Vaughn Road
Suite 215
Kennesaw, GA 30144 USA
Please remember to include your invoice number or account number on
your check when mailing.
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IPDID Local Origination Service |
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What locations does IP Communications
offer local numbers (DIDs)? |
You can find a list of our current
coverage area on our web site
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How do I view your coverage area? |
You can find a list of our current
coverage area on our web site
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What is a DID? |
Short for direct inward dialing (also
known as direct dialing inward), a service of an LEC or local phone
company that allows an organization to have numerous individual
phone numbers for each person or workstation in its PBX system that
run off of a small block of dedicated telephone numbers. DID allows
the multiple lines to be connected to the PBX all at once without
requiring each to have a physical line connecting to the PBX.
For example, if an organization has 25 employees and each employee
has a separate telephone number, or extension, within its physical
location, the organization can rent 10 physical trunk lines from the
telephone company that will allow 10 phone calls to take place
simultaneously. Others would have to wait for an available line and
anyone dialing into the system while all 10 lines are in use would
get either a busy signal or be channeled into a voice mail system. A
DID system does not require a PBX operator and can be used for fax
and voice transmissions.
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Is there a limit to the number of calls
I can place on across my trunks or phone numbers? |
No. There are no usage limits
across IPDID trunks or numbers. One port can handle one
concurrent call. You can have many numbers assigned to a
port or you can have one number assigned to many ports.
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Can
I choose my numbers ? |
No, this is currently not supported.
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Does IP Communications support 911? |
Yes. With qualifying services,
you can receive 911 service.
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How many simultaneous incoming calls
can a single number support? |
No. There are no usage limits
across IPDID trunks or numbers. One port can handle one
concurrent call. You can have many numbers assigned to a
port or you can have one number assigned to many ports.
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Can I add additional ports at any time? |
Yes. With most services, you can
add numbers or ports to your account at any time.
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What does "Unlimited Incoming Calls"
mean? |
Unlike most DID providers, IP
Communications does not restrict its usage to home users or low
usage callers. Unlimited means unlimited. Place as many
calls as you like across your IPDID trunks.
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Is there a limit to the number of ports
I can purchase? |
No. There are no limits to the
number of numbers or ports you can order. However, for
extremely large orders, we do ask that you give us some prior
notice.
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Can I place outbound calls via the IP
Communications network ? |
Yes. Simply visit
www.talkinip.net to signup
for pay-as-you-go outbound services.
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I do not have a voip gateway, can IP
Communications provide one? |
Yes. Just contact a sales
representative for a list of our VoIP devices.
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Can you route my IPDID or IP800 number
to a PSTN phone (a cell phone for example)? |
Yes. If you do not have a
VoIP device, we can alternatively point them to a PSTN (public
switched telephone number) you provide* (e.g. your office number
for). There is a charge for the PSTN leg of the call.
For example if you pointed your DID to a cell phone in Chicago, your
inbound DID leg of the call would not be charged, but the
termination to the cell phone would be charged at the current US
outbound rate.
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I want to transfer my current number to
IP Communications. What do I do? |
Let us know
what number you wish to have transferred, and we will check to see
if that number is transferable.
Once we have verified that your phone number is transferable, simply
download and fill-out our Local Number Porting form
detailing your name, service address, and billing telephone number
and return it to us to continue your transfer request.
Once the request has been processed, you will be notified via email
of all status changes.
More information on transferring your number
to IP Communications.
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How much does it cost to add a phone
number to my service? |
Our rates can be
viewed on our site at (http://www.ipcomms.net/html/product-ipdid.html)
or a sales associate can be contacted to further explain all the
details of the service.
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How much does it cost to add an
additional port to my account? |
Our rates can be
viewed on our site at (http://www.ipcomms.net/html/product-ipdid.html)
or a sales associate can be contacted to further explain all the
details of the service.
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How much is the Business DID package? |
Our rates can be
viewed on our site at (http://www.ipcomms.net/html/product-ipdid.html)
or a sales associate can be contacted to further explain all the
details of the service.
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Does IP Communications provide volume
discounts? |
On certain services we are able to
offer reseller or volume discounts. Please contact a customer
service and ask about our volume rates.
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Does IPDID service have setup fees? |
For most services, there are no setup
fees.
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What are the rates for IP
Communication’s DID service? |
Our rates can be
viewed on our site at (http://www.ipcomms.net/html/product-ipdid.html)
or a sales associate can be contacted to further explain all the
details of the service.
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What is a Toll Free Number? |
Toll-free
numbers are numbers that begin with one of the following three-digit
codes: 800, 888, 877, or 866. Toll-free numbers allow callers to
reach businesses and/or individuals without being charged for the
call. The charge for using a toll-free number is paid by the called
party (the toll-free subscriber) instead of the calling party.
Toll-free numbers can be dialed directly to your business or
personal telephone line.
Toll-free numbers are very common and have proven successful for
businesses, particularly in the areas of customer service and
telemarketing. Toll-free service provides potential customers and
others with a “free” and convenient way to contact businesses.
Toll-free numbers are also increasingly popular for personal use.
For example, parents can obtain toll-free numbers to give to a young
adult who is away at college, allowing that young adult to call home
anytime without having to make a collect call or pay for the call.
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How do toll-free numbers
work? |
A toll
free number just forwards to or points to a regular local number. No
special equipment or additional line or installation is required.
When a call is placed to a toll free number, the Local Exchange
Company (LEC) queries the SMS/800 Database to determine the
inter-exchange carrier (long distance company) responsible for
carrying the call. The inter-exchange carrier then picks up the
call, applies the appropriate features or routing, creates a call
record for billing, and routes the call to the terminating number,
trunk ID or circuit ID to which the toll free number is programmed
to ring.
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After signing up for a free account, how does the free DID forward
to my IP phone if my IP changes? |
Simply log in to AMI (www.myipcomms.net)
and submit a request to have your IP changed.
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If a company has an 800 number, can I also reach them by calling the
888, 877, or 866 equivalent? |
No. Toll
free codes are separate and distinct. Different companies may have
the same phone number, but with different toll free codes.
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Can I
use more than 7 digits for a toll free number? |
Phone
numbers have 7 digits, so although you can not use less than 7
digits, you can use more. The only exception is that if callers dial
8 digits on their cell phone it may not go through. And if you have
a Z near the end of your name or a part of your name that is
difficult to spell it may be a good idea to push that off the end of
the vanity number. But in general, using 7 digits will give you
greater flexibility when searching for a toll free number by
providing more options from which to choose.
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How are toll-free numbers assigned to subscribers? How can I get a
toll-free number? |
Toll-free
numbers are usually assigned on a first-come, first-served basis.
Entities called Responsible Organizations ("RespOrgs"), which are
usually toll-free service providers or carriers, have access to a
database that contains information regarding the status of all
toll-free numbers.
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I've heard that toll-free numbers are portable; what does that mean? |
Portability means that toll-free subscribers can change carriers
without having to obtain a new toll-free number. Subscribers may
also change Responsible Organizations if they choose to do so.
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What is the FCC's role in the market for toll-free services? |
The
Commission regulates or sets the rules under which toll-free numbers
can be used or obtained. The Commission is not involved in the
day-to-day allocation of toll-free numbers and does not have access
to the toll-free database.
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Toll-Free Codes -
800, 888, 877, 866 |
Today,
there are four toll-free codes: 800, 888, 877, and 866. Although
800, 888, 877, and 866 are all toll-free codes, they are not
interchangeable. 1-800-234-5678 is not the same number as
1-888-234-5678. Calls to each toll-free number are routed to a
particular local telephone number.
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Toll-Free Directory
Assistance |
Toll-free
directory assistance for some toll-free numbers can be obtained by
calling 1-800-555-1212. The service is free. Not all toll-free
numbers are listed – only the numbers for subscribers that choose to
list them. The Federal Communications Commission (FCC) plans to
address how to promote competition among multiple providers of
directory assistance, including directory assistance for toll-free
numbers. In the meantime, 1-888-555-XXXX numbers are not being
assigned to subscribers.
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Has the FCC Commission issued any rulemakings regarding toll-free
numbers? |
Yes. On
October 5, 1995, the Commission released a Notice of Proposed
Rulemaking (Toll-free Service Access Codes, Notice of Proposed
Rulemaking, FCC Rcd 10 13962 (released October 5, 1995)) to address
issues regarding the efficient, fair, and equitable allocation of
toll-free numbers. Subsequent to the Notice of Proposed Rulemaking,
the Common Carrier Bureau, acting on delegated authority, issued a
Report and Order (Toll-free Service Access Codes, Report and Order,
11 FCC Rcd 2496 (released January 25, 1996)) that addressed those
issues crucial to the opening of the 888 code for toll-free calling.
On April 11, 1997, the Commission released a Second Report and Order
addressing issues pertaining to the efficient, fair, and equitable
allocation of toll-free numbers. On October 9, 1997, the Commission
released a Third Report and Order addressing issues relating to toll
free database administration. On March 31, 1998, the Commission
released a Fourth Report and Order ( erratum ) addressing the issue
of vanity-number assignment. Some issues raised in the Notice of
Proposed Rulemaking remain unaddressed, and the proceeding is still
open. On July 5, 2000, the Commission released a Fifth Report and
Order in the matter of Toll Free Service Access Codes, Database
Services Management, Inc.'s Petition for Declaratory Ruling, and
Beehive Telephone Company's Petition for Declaratory Ruling. CC
Docket No. 95-155, NSD File No. L-99-87, NSD File No. L-99-88.
Custom Toll Free makes sure your business doesn't get caught up in
all the legal jargon. We help you decide on the numbers that really
drive your business, find it for you, and get it working for you.
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What Is A
"Vanity" Number and How Can I Get One? |
A “vanity”
number is a toll-free telephone number that also spells a person’s
or company’s name or spells a word or acronym that is chosen by the
subscriber, such as 1-800-FLOWERS or 1-888-NEW-CARS. To find out
whether a specific toll-free number is available, contact any
RespOrg or toll-free service provider.
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"Warehousing/Hoarding" Toll-Free Numbers |
“Warehousing” by toll-free service providers is prohibited by the
FCC’s rules. A toll-free service provider may not legally reserve a
toll-free number without having an actual toll-free subscriber for
whom the number is being reserved. RespOrgs or toll-free service
providers who warehouse numbers are subject to penalties.
“Hoarding” by subscribers is similarly prohibited and illegal. A
subscriber may not acquire more toll-free numbers than the
subscriber intends to use. Hoarding also includes “number brokering”
– it is illegal for a subscriber to sell a toll-free number for a
fee.
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How do I signup for
IP800 service? |
Getting started with IP Communications is easy. Simply visit
our website at www.ipcomms.net, and choose the service and package
you desire. Simply choose the phone numbers or number
locations you require and complete the order process. Within
24 hours (usually less) of receiving your order, you will receive a
service provisioning letter that will contain all the information
you need to configure your VoIP device. Configure your device
with our information and begin placing and/or receiving calls.
You will be able to monitor and maintain your service online using
AMI our Account Management Interface found here:
www.myipcomms.net.
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Can I buy a
toll free number to receive calls? |
Yes.
Check out our IP800 Toll Free service online.
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Does IP Communications have to supply me with a new number or can I
keep my existing toll free number? |
Most
numbers can be transferred to IP Communications, however there are
some restrictions:
-
Your
current account must be active and in good standing with your
existing provider.
-
There may not be a
line freeze or a pending order on your existing phone.
-
The number you are
transferring must be within IP Communications’ service area.
You will need to provide
IP Communications with the exact Name, Service Address, and Billing
Telephone Number (BTN) on record with your current provider,
including any capitalization, punctuation and/or abbreviations.
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Do I need to also subscribe to your IPDID service in order to get
the IP800 service? |
No.
You can order IP800 Service by itself.
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What is a Payphone Surcharge, and why am I being charged for it? |
Toll-free
calls, including calls billed to calling cards or credit cards, also
do not require a coin. The Communications Act, however, requires the
FCC to establish a per-call compensation plan to ensure that all
payphone service providers (PSPs) are fairly compensated for every
completed intrastate and interstate call using their payphones --
except for emergency calls. The toll-free number provider, calling
card service, or credit card company generally pays this
compensation, but they may pass this cost on to users in the rates
charged.
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How are my minutes of
use rounded? |
Billing
increments for all usage based services (except Mexico) are 30/6.
Thirty seconds minumum and 6 second increments there after.
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Are there
any monthly minimums for IP800? |
Yes.
Check out our IP800 Toll Free service online.
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Are RESPORG per-number service charges charged monthly or is it a
one-time charge? |
The charge
is a one-time charge and is charged per number transferred.
Currently it is $10 per number transferred. However this is
subject to change so pleae check with your IP Communications sales
representative for details.
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Number Transfer (LNP/RespOrg) |
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What is LNP (local
number portability)? |
Under the Federal Communications Commission’s
(FCC’s) “local number portability” (LNP) rules, so long as you
remain in the same geographic area, you can switch telephone service
providers, including interconnected Voice over Internet Protocol
(VoIP) providers, and keep your existing phone number. If you are
moving from one geographic area to another, however, you may not be
able to take your number with you. Therefore, subscribers remaining
in the same geographic area can now switch from a wireless,
wireline, or VoIP provider to any other wireless, wireline, or VoIP
provider and still keep their existing phone numbers.
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I want to transfer my current number to IP
Communications. What do I do? |
To begin, let us know what number you wish to
have transferred, and we will check to see if that number is
transferable.
Once we have verified that your phone number is transferable, simply
download and fill-out our Local Number Porting form
detailing your name, service address, and billing telephone number
and return it to us to continue your transfer request.
Once the request has been processed, you will be notified via email
of all status changes.
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Does it cost anything to transfer my current number to IP
Communications? |
Yes. Currently it is $15 per number
transferred. However this is subject to change so pleae check
with your IP Communications sales representative for details.
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Are there any restrictions in transferring my
number? |
Most numbers can be transferred to IP
Communications, however there are some restrictions:
Your current account must be active and in good standing with your
existing provider. There may not be a line freeze or a pending order
on your existing phone.
You will need to provide IP Communications with the exact Name,
Service Address, and Billing Telephone Number (BTN) on record with
your current provider, including any capitalization, punctuation
and/or abbreviations.
The number you are transferring must be within IP Communications’
service area.
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What
happens if my number can’t be transferred? |
If IP Communications cannot transfer your
number you may elect to get a new number in your area or elsewhere.
There are several reasons that may cause a failure to your transfer
request. Here are some more common problems:
Number is not within IP Communications service area for transfers.
Name and Address mismatch or invalid
Your number information must match exactly what your previous
carrier has on their Record. If it does not match exactly, they will
reject our request.
Billing Telephone Number is incorrect
Your Billing Telephone Number must match exactly what your previous
carrier has on their bill. If it does not match exactly, they will
reject our request.
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Are LNP service charges charged monthly or is it a one-time charge? |
The charge is a one-time charge and is charged
per number transferred. Currently it is $15 per Local Number (LNP) and $10 per toll free and $10 per Vanity per number transferred.
However this is subject to change so pleae check with your IP
Communications sales representative for details.
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How do I view all call detail records for a
specific billing period? |
Yes.
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To view call detail records for an entire billing period, you
follow these steps:
Log into AMI
1. Click
on "billing" in the
AMI menu. This will expand the billing
section.
2.
Next, click on the "invoice history" option
3.
Here you will find a list of your last 3 invoices for you account
and the corresponding call detail records for each billing period.
Click "Download" for the period in which you wish to see call
details.
4.
It may take several minutes to download...message will appear.
5.
Click the "Click here to download" link and choose "Save As..." to
save the Call Detail Records to your computer.
6.
The "Do you want to open or save this file" box will appear.
Choose "Save As..." to save the Call Detail Records to your
computer.
The downloaded file will be in .csv format.
If you use Microsoft Excel to view this data, please remember that
only the first 65,000 records will be displayed. To view more
than 65,000 records, you will need to use another application.
This is a limitation of Microsoft Excel, and not of the downloaded
file.
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How can I view CDRs (Call Detail Records) for specific dates I
specify? |
To view call detail records for an entire
billing period, you follow these steps:
Log Into AMI (www.myipcomms.net)
Click on Reports
then traffic stats in the
AMI menu.
Next, the Traffic
Stats tool will appear. From here you can specify the dates in
which you wish to view call detail records.
Here you will find a
list of your last 3 invoices for you account and the corresponding
call detail records for each billing period. Click Download
for the period in which you wish to see call details.
4.
It may take several minutes to download...message will appear.
5.
Click the "Click here to download" link and choose "Save As..." to
save the Call Detail Records to your computer.
6.
The "Do you want to open or save this file" box will appear.
Choose "Save As..." to save the Call Detail Records to your
computer.
The downloaded file will be in .csv format.
If you use Microsoft Excel to view this data, please remember that
only the first 65,000 records will be displayed. To view more
than 65,000 records, you will need to use another application.
This is a limitation of Microsoft Excel, and not of the downloaded
file.
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How do I add
new phone numbers to my account? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
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How do
I remove an existing number from my account? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
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Where can I view a list of my current rates and service summaries? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
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|
Where can I view
my most current invoice? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I make an online
payment? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
Where can I view my
invoice history? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I sign up for Autopay? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I add ports to my
account? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
Can I view my order
history online? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I view
my current trunk configuration? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I
see a list of all of my phone numbers? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do
I see a list of all of my service locations? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to
top |
|
How do I report an issue with my IP Communications service? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
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top |
|
Can I chat with a customer service representative online? |
Yes. You can chat with customer service
using any of the Click to Chat buttons on our website and in the
Account Management System.
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|
I would like to report a problem with the online account management
interface AMI. |
You can report issues with our website and AMI
(account management system) to
webmaster@ipcomms.net
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|
I forgot my online account management interface password or
username. |
If you have lost your AMI username and
password, you can simply go to our "Recover Your Password" page and
recover it.
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|
Questions about FREE DIDs |
| |
|
How can I try or test
the service? |
We are so sure that you will be 100% satisfied
with our service, we are giving away one free number to show it.
With our FREE-DID offer you will receive one U.S. local number from
one of our over 5000 service locations along with 2 free ports.
We will deliver this number to your IP enabled device and allow
unlimited incoming calls at NO CHARGE.
back
to top |
|
How do Free DIDs Work? |
Simply complete the number request form, and
we will quickly provide you with your free number.
http://www.ipcomms.net/html/freedid.html
Enjoy free, unlimited calling - our treat!!!
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to top |
|
Are there any
limitations with FREE DIDs? |
Terms:
Activation FREE DIDs are at the sole discretion of IP
Communications. IP Communications reserves the right to refuse or
deactivate service at any time. In order for your number to
remain active, at least one call to your number must be received per
month. Limit one free number per customer. Offer not
valid to existing or prior customers of IP Communications, LLC.
Free numbers are offered as a "best-effort" service. IP
Communications does not offer technical support with free numbers.
http://www.ipcomms.net/html/FREEDID_Landing2.htm
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to top |
|
How long can I keep my
FREE DID? |
As long as you like. It's yours.
We only ask that you make at least one call across your number per
month, to let us know that you are still using it.
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|
How many free did
numbers can I receive? |
Limit one FREE DID per customer. If you
need more, please see our IPDID service starting at only $9.99 per
month.
https://www.myipcomms.net/oop/orderipdid/default.asp
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to top |
|
Are there any per minute
charges? |
No. There are no per minute charges for
FREE DIDs.
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top |
|
How many ports
does my FREE DID come with? |
FREE DIDs come with 1 Number & 2 Ports.
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top |
|
Do I
have to purchase something to get a FREE DID? |
No purchase necessary.
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|
Why are you giving away FREE DIDs? What's the catch? |
We are so sure that you will be 100% satisfied
with our service, we are giving away one free number to show it.
With our FREE-DID offer you will receive one U.S. local number from
one of our over 5000 service locations along with 2 free ports.
We will deliver this number to your IP enabled device and allow
unlimited incoming calls at NO CHARGE. No gimmics, no catch!
Simply complete the number request form, and one of our service
representatives will contact you and provide you with your free
number.
Enjoy free, unlimited calling - our treat!!!
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to top |
|
Can I choose
where I want my number from? |
With FREE DIDs we assign you a number from our
inventory. If you would like to choose your own number
location, please see our IPDID service starting at only $9.99 per
month.
https://www.myipcomms.net/oop/orderipdid/default.asp
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to top |
|
Do FREE DIDs support H.323 |
No. FREE DIDs are SIP only.
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|
Can I add
additonal ports to my FREE DID? |
No. If you require more ports, please
see our IPDID service starting at only $9.99 per month.
https://www.myipcomms.net/oop/orderipdid/default.asp
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to top |
|
Can I
make outbound calls on my FREE DID account? |
No. For outbound calling please visit
our pay as you go service
www.talkinip.net .
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to top |
|
Can I get a free
DID outside of the USA? |
No. US locations only at this time.
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top |
|
Are there
any terms or conditions with FREE DIDs? |
Terms:
Activation FREE DIDs are at the sole discretion of IP
Communications. IP Communications reserves the right to refuse or
deactivate service at any time. In order for your number to
remain active, at least one call to your number must be received per
month. Limit one free number per customer. Offer not
valid to existing or prior customers of IP Communications, LLC.
Free numbers are offered as a "best-effort" service. IP
Communications does not offer technical support with free numbers.
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to top |
|
Resellers and Agent Questions |
| |
|
Service Rate and Fee Questions |
| |
|
Can I pay via Paypal ? |
Yes. You can send payments to PayPal address:
billing@ipcomms.net
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I want my credit card to be charged automatically every month, is
that possible ?
|
Yes. Simply login to AMI
(http://www.myipcomms.net ) and signup for APP (Auto Payment Plan).
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|
Do
you debit my credit card automatically every month ? |
It depends on the service you signed up for.
In most cases, your credit card will be charged automatically for
bills.
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to top |
|
I do not want my credit card to be charged automatically every
month, is that possible ? |
Yes. Contact a member of customer
service and if you qualify, you will not be charged automatically
every month.
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to top |
|
Where can I find information on inbound rates and plan fees? |
Rates and plan costs can be found via our home
page. Just select the product or plan you desire.
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to top |
|
What are your long
distance rates |
Our outbound rates can be found at our
TalkinIP Website
www.talkinip.net
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|
Where can I find information on long-distance and international
outbound calling? |
TalkinIP is our Pay-as-you-Go outbound
service.
www.talkinip.net
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|
Are
DIDs calculated monthly or is it a one-time fee? |
All phone numbers are billed for monthly.
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to top |
|
Are port fees calculated monthly or is it a one-time fee? |
All port fees are monthly.
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to top |
|
How often am I billed? |
Bills are delivered via Email by the 5th of
each month.
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to top |
|
Can I cancel my
service at any time? |
There are no fees to cancel your account.
IP Communications only requires thirty (30) days notice in order to
cancel an account. The 30 day notice period will begin at the time
of receipt of the Account Cancellation Form by IP Communications.
You can request an Account Cancellation Form by contacting IP
Communications customer support.
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|
Are there any cancellation
fees? |
There are no fees to cancel your account.
IP Communications only requires thirty (30) days notice in order to
cancel an account. The 30 day notice period will begin at the time
of receipt of the Account Cancellation Form by IP Communications.
You can request an Account Cancellation Form by contacting IP
Communications customer support.
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|
What is USF (universal service fund) and why am I being charged for
it? |
This is a charge to recover the amount
telecommunications providers must contribute to the Federal
Universal Service Fund, which helps keep local phone rates
affordable
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to top |
|
Can I view my calls online? |
Yes. You can view and manage all of your
account information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
back to top |
|
How do I
order new number, ports or services? |
You can view and manage all of your account
information with AMI our Account Management Interface.
AMI gives you quick and easy access to your IP Communications
services and account information 24 hours a day, 7 days a week.
With a simple click of your mouse, you can:
-View or change
account details
-Download call
detail records
-Manage voice
services
-Update technical
information
-View and pay bills
-Order new services
-Open a support
ticket
Registration is fast and easy. Just follow these 3 simple steps:
1.Visit
http://www.myipcomms.net
and click on the link that reads:
"Or, if this is your first time here, register online. "
2.Fill out the registration information.
3.Verify your confirmation email. Sign in and enjoy!
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|
Do you pro-rate my bill if I sign up for service or add new services
in the middle of the month. |
Yes.
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|
How
do I pre-pay for outbound and international calls? |
Simply visit
www.talkinip.net
and login to your pre-paid outbound account and submit payment.
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top |
|
What happens
when my calling account is empty? |
When you are out of credit on your pre-paid
outbound account (www.talkinip.net
), you should add more money to your account right away. If you
exceed your allotted minutes, you will no longer be able to place
calls until you add money to your account (but you can still receive
calls) if you also have an inbound service with IP Communications.
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top |
|
How do I fund my account? |
Online: Log In to your account online and
recharge using a credit card or major debit card.
By phone: Add funds with credit card, PayPal, or check by calling
1-800-678-1475 or contact a customer service representative.
For those clients with larger-volume voice transit needs, you may
also fund your account via bank wire-transfer.
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|
What type of
service does TalkinIP provide? |
TalkinIP provides quality transit of SIP based
phone calls to Tier-1 telecom carriers. There are no monthly rates
and no contracts... we are a "Pay-As-You-Go" provider.
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top |
|
Is
your service an inbound service, outbound service or both? |
TalkinIP offers speedy transit of your voice
calls to regular destinations on the Publicly Switched Telephone
Network (PSTN). We provide inbound (local and toll free) services
through our IPDID and IP800 products.
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|
Is
there a limit to the number of simultaneous calls I can place? |
No. The amount of simultaneous calls that can
be made is dependent on the balance in your account. At the
beginning of each call, up to two-hours of credit at that calling
rate is held in-escrow by our system. A second and simultaneous call
will attempt to hold another two-hours of credit in-escrow and so
forth. At the conclusion of each call, any remaining credit is
released back to your account for use.
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|
I'm only a small business/home user, can I still use your service? |
We realize that the small-to-medium business
market and hobby users are important clients and we treat them all
the same. Our systems have been designed for both large and hobby
users alike.
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top |
|
How
do I start using your service? |
Getting started is easy. Simply register,
place funds in your account, configure your equipment and start
sending calls. You can start with as little as $15.00 USD to try our
services.
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top |
|
Does IP Communications offer competitive international outbound
calling? |
TalkinIP enables you to make calls to locations worldwide at the
absolute lowest rates. You can use software on your computer or a
SIP enabled device (Asterisk, Trixbox, SIP Gateways, IAD, IP
PBX) to place calls using the internet.
TalkinIP provides you outbound calling to any destination within the
continental USA and service to over
227 countries at very competitive international rates. Get
started for as little as $15 prepaid credit. Your credit will
never expire, so you can make calls next week or next year.You receive USA 48 States, Canada, and Toll-Free termination for only $0.02 / minute
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top |
|
Do you offer
service to A-Z destinations? |
We have the ability to deliver your call to
virtually any location worldwide. For the security of our clients,
we do restrict access to a handful of premium destinations including
but not limited to satellite connections (INMARSAT) and pay-per-use
entertainment services.
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|
What
equipment do I need to use with TalkinIP? |
Clients requiring the most flexibility may use
any SIP enabled device they choose. We also support open
source based IP PBXs such as Asterisk, TrixBox or FreeSwitch. For
those clients who require less frequent use of our service they may
opt for a SIP ATA adaptor, which connects their regular telephone to
our service.
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top |
|
What if I don't have Asterisk or SIP hardware, can I use a software
phone on my PC? |
Yes. Our service allows the use of a PC "soft
phone". There are several soft phones available on the market.
We recommend both X-Lite and Zoiper for their simplicity of
use. Instructions for downloading and installing these and
other SIP devices can be found here. You will require a headset or
microphone to use soft phones.
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top |
|
What protocols does
your service use? |
TalkinIP primarily uses SIP. Clients
using SIP hardware or telephones, Asterisk or related software may
utilize either SIP. Other protocols may be supported depending
on the requirements. Contact an IP Communications customer
support representative for further information.
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| What
audio codecs do you support? |
IP Communications supports the use of G.729
and G.711u codecs, and we offer them in that order. Other
codecs may be supported depending on the requirements.
G.711
G.711 is an ITU-T standard for audio commanding. It is primarily
used in telephony. The standard was released for usage in 1972.
G.711 is a standard to represent 8 bit compressed pulse code
modulation (PCM) samples for signals of voice frequencies, sampled
at the rate of 8000 samples/second. G.711 encoder will create a 64
kbit/s bit stream.
There are two main algorithms defined in the standard, mu-law
algorithm (used in North America & Japan) and a-law algorithm (used
in Europe and the rest of the world). Both are logarithmic, but the
later a-law was specifically designed to be simpler for a computer
to process. The standard also defines a sequence of repeating code
values which defines the power level of 0 dB.
The equations are:
mu-law:
y = ln(1 + ux) / ln(1 + u) with u = 255
A-law:
y = Ax / (1 + ln A) for x <= 1/A where A = 87.6
y = (1 + ln Ax) / (1 + ln A) for 1/A <= x <= 1
G.729
G.729 is an audio data compression algorithm for voice that
compresses voice audio in chunks of 10 milliseconds. Music or tones
such as DTMF or fax tones cannot be transported reliably with this
codec, and thus use G.711 or out-of-band methods to transport these
signals.
G.729 is mostly used in Voice over IP (VoIP) applications for its
low bandwidth requirement. Standard G.729 operates at 8 kbit/s, but
there are extensions, which provide also 6.4 kbit/s and 11.8 kbit/s
rates for marginally worse and better speech quality respectively.
Also very common is G.729a which is compatible with G.729, but
requires less computation. This lower complexity is not free since
speech quality is marginally worsened.
The annex B of G.729 is a silence compression scheme, which has a
VAD module which is used to detect voice activity, speech or non
speech. It also includes a DTX module which decides on updating the
background noise parameters for non speech (noisy frames). These
frames which are transmitted to update the background noise
parameters are called SID frames. A comfort noise generator (CNG) is
also there, because in a communication channel, if transmission is
stopped, because it's not speech, then the other side may assume
that link has been cut. This is also taken care of by the annex B
standard.
Contact an IP Communications customer support representative for
further information.
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top |
| What is the format of my outbound dial pattern (dialing string)? |
When sending calls to IP Communications, it is important that you format calls sent from your VoIP device correctly. When sending outbound calls to IP Communications, there should be no preceding characters (plus signs "+" for example) in the number you are sending to us. Instead, your device should only provide the country code and the number to be dialed as one dial pattern (dial string).For example if the number you are dialing is in the united states (country code 1) and the number you wish to dial is 6784601475, your Dial Pattern should be as follows:
Your Dialing Pattern = 16784601475
(This is correct!)
Your Dialing Pattern = +16784601475 (This is NOT correct!)
There should be no plus sign (+) in front of the country code. If a plus sign is sent before the number, it will not accepted for outbound calling.
Common mistakes:
- Phone number sent with a plus sign (+) before the country code
- No country code sent before the phone number
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top |
|
Technical Support - Asterisk |
| |
|
Does the
service support IAX, Asterisk, IAX2? |
Yes. We support the IAX protocol.
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to top |
|
Can I use
Asterisk or trixbox with your service? |
Yes. We support all of the most popular
flavors of Asterisk Open Source PBXs
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|
How do I configure my asterisk pbx to work with your service with
codec G.711? |
Below are some configuration examples for the
open source PBX Asterisk TM . These examples may vary depending on
your software versions, implementation type, and more. These
are examples taken from most common customer configurations.
SIP Trunk Sample Entry - Inbound Service - g711ulaw
(Listed below is an example of what you would put in your SIP Trunk
entry.)
Create a SIP trunk for g711ulaw.
Trunk Name: ipcomms
PEER Details:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend
USER Context: from-pstn
USER Details:
allow=ulaw
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend
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|
How do I configure my asterisk pbx to work with your service with
codec G.729? |
(Listed below is an example of what you would
put in your SIP Trunk entry.)
Create a SIP trunk for g729.
Trunk Name: ipcomms
PEER Details:
allow=g729
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend
USER Context: from-pstn
USER Details:
allow=g729
context=from-pstn
dtmfmode=rfc2833
host=64.154.41.100
insecure=very
nat=yes
type=friend
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|
How do I configure my Asterisk PBX to work with your service if I
have a private IP Address? |
What do I do if my asterisk server has a
private IP Address?
Please enter the following two settings (externip and localnet) at
the top of your sip.conf file in the general section.
If public, nothing needs to be added to the general section.
Depending on your setup, you might need to forward all ports
to your Asterisk server's private IP address or put the private IP
address of the Asterisk server on the DMZ.
externip=WAN IP address
localnet=192.168.1.0/255.255.255.0 (or whatever your private LAN is
configured for)
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|
Do you provide a SIP username and password for registration or do I
have to have a static IP address? |
Yes, we do provide a SIP username and password
registration option during the online ordering process.
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|
I am new to asterisk. Can you help me with my setup and
configuration? |
Free Basic Support:
Everyone needs a little help some time. Why should you have to
pay for it?
When you sign up for any of our business packages, you will receive
free basic asterisk support. We will assist you in basic
asterisk setup and configuration with our service to get your
service up and running quickly.
IP Communications has designed products and services that are
designed to meet the needs of your asterisk PBX. Our network
can support SIP and IAX trunk connectivity. We have online
support documents and configuration examples available to guide you
through your asterisk setup and configuration.
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|
What
type of server do I need to install asterisk? |
Generally, the type of pc required to run
asterisk depends on the type of coding you will be doing. If
you are not converting codec types (meaning receiving a call G.711
and sending it back out G.729, then you will need less of a pc than
if you will be doing lots of transcoding) As a good start, we
recommend that you should run asterisk on a dedicated machine,
preferably 2.4ghz or faster with 512mb of RAM or more. .
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|
How many users does
Asterisk support? |
Lots, but it is not a service provider, its
more geared towards office use, and there are many factors affecting
usable size.
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to top |
|
Where can I download the most recent version of Asterisk? |
You can download asterisk at the following
website.
http://www.asterisk.org/downloads
back
to top |
| Where
can I find instructional videos for Asterisk? |
http://www.asterisktutorials.com/
back to top |
How do i fix one-way audio on my Asterisk PBX? |
The following is used to fix one way audio on your Asterisk PBX if you do not have a public static IP address on your asterisk pbx, and you are using IP authentication (not SIP username/password Registration).
You will need to enter the following two settings (externip and localnet) at the top of your sip.conf file in the general section.
externip=WAN IP address
localnet=192.168.1.0/255.255.255.0 (or whatever your private LAN is configured for)
Depending on your setup, you might need to forward all ports
to your Asterisk server's private IP address or put the private IP address of the Asterisk server on the DMZ.
Note: If you have a public static IP address assigned to the server itself, then you do not need to add these settings to your config. |
|
Technical Support - General |
| |
|
Do
you have a sample configuration for a Cisco gateway? |
Below are some configuration examples for
Cisco gateways. These examples may vary depending on your software
versions, implementation type, and more. These are examples
taken from most common customer configuration. Depending on your
gateway version some commands may differ.
Sample VoIP Entry - H323 - Outbound Service
Sample POTs Entry - Inbound Service
!
dial-peer voice 6781475 pots
description IPC
huntstop
destination-pattern 6784601475
no digit-strip
port 0:D
!
dial-peer voice 1234 voip
description IPC
huntstop
destination-pattern . or .T
progress_ind setup enable 3
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g729r8
tech-prefix yyyyyyy
!
xxx.xxx.xxx.xxx is the IP address of IPC's destination gateway.
yyyyyyy is the
prefix assigned to your account.
Sample VoIP Entry - SIP - Outbound Service
!
dial-peer voice 1234 voip
description IPC
huntstop
destination-pattern yyyyyyy. or yyyyyyy.T
progress_ind setup enable 3
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay rtp-nte h245-signal h245-alphanumeric
codec g729r8
!
xxx.xxx.xxx.xxx is the IP address of IPC's destination gateway.
yyyyyyy is the prefix assigned to your account.
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to top
|
|
Do you
have a sample config for x-Lite softphone? |
X-Lite Configuration Instructions
back to top
These instructions are for use with SIP registration. This
example will not work with IP address registration (Free DIDs for
example will not use this config).
STEP 1:
Download the soft phone
here
and follow the prompts from the install wizard. Click Finish to complete
the installation.
STEP 2: Launch
the soft phone and wait for a window to pop up with your "SIP Accounts"
and click the add button
STEP 3:
Next the "Properties of
Account 1" window will pop up. Place the following information there:
1.
Display Name
: Any name you
wish to place here
2.
User Name
: Enter the
‘Username’ we sent you via your Sign Up confirmation email
3.
Password
: Type the
password we assigned you on your Sign up confirmation email
4.
Authorization user name
: Enter the ‘Username’ we sent you via your Sign Up
confirmation email
5.
Domain:
Enter the following domain name "
sip.ipcomms.net"
Troubleshooting
If you get one-way
audio, or cannot register you are probably behind NAT or firewall is
enabled on your PC. Disable firewall on your PC and make the following
changes.
- Go to "Sip Account
Settings.." on the softphone then go to properties
- Click on "Topology"
tab and choose "Use local IP address" under "IP Address"
- Click "Enable ICE"
option and then click on "OK" button.
Restart the
application and try making calls again.
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|
|
Do
you have any sample configs for ZoIPer Softphones?
 |
ZoIPer Configuration Instructions
back to top
STEP 1:
Download the Softphone
here
and follow the prompts from the install wizard. Click Finish to complete
the installation.
STEP 2: Launch
the softphone and wait for the main window to pop up.
STEP 3:
Right click anywhere on the main window and click options.
STEP 4:
Click
Add new Sip account
STEP 5:
Enter
your name in the pop up window and click OK
STEP 6:
Next the "SIP Account
Options" window will pop up. Place the following information there :
1.
Domain:
Enter the following domain name "
sip.ipcomms.net"
2.
User Name
: Enter the
‘Username’ we sent you via your Sign Up confirmation email
3.
Password
: Type the
password we assigned you on your Sign up confirmation email
4.
Caller ID Name
: Enter the ‘Username’
we sent you via your Sign Up confirmation email
STEP 7:
Click Ok
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|
|
Do you have a sample config for fring (so I can make sip calls with
your windows Mobile PDA or phone)?
 |
Make SIP Calls with your Windows Mobile PDA or Phone
fring
leverages the internet connectivity traditionally used for mobile email
retrieval and web browsing to provide mobile VoIP communications so you
can talk and instant message for free! fring allows you to make
calls over the Internet using any windows mobile phone.
|
 |
fill in your details
Complete your details & click "join". |
|
|
|
 |
download fring
On your handset, click on the link within the SMS to
automatically
download the fring application. |
|
|
|
 |
install & register
Follow the onscreen instructions on your handset to install and
register. |
|
|
What codecs do you support? |
Depending on your specific requirements, we
support several delivery methods (VoIP protocols). Below is a
list of our supported delivery methods, and the associated Codec
each supports.
SIP (g729, g711ulaw)
H323 (g729, g711ulaw)
IAX (g711ulaw)
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|
|
How do I contact
technical support |
Contact Technical Support
Engineering Questions
Email:
noc@ipcomms.net
or you may open an Online Support Ticket
Order Fulfillment Questions
1-800-228-8596
1-678-460-3797
3. Please click here to update your technical information and
specify your routing requirements.
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|
|
What codecs do you support? |
Depending on your specific requirements, we
support several delivery methods (VoIP protocols). Below is a
list of our supported delivery methods, and the associated Codec
each supports.
SIP (g729, g711ulaw)
H323 (g729, g711ulaw)
IAX (g711ulaw)
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|
|
Do you
support Automatic PSTN or IP failover? |
Yes. Automatic PSTN or IP failover is
available upon request.
back to
top |
|
Do you support e911? |
Yes, IP Communications does provide e911 with
certain services. IP Communications 911 service operates
differently than traditional 911. Ask your IP
Communications service representative for more details.
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|
Do you support T.38 Faxing |
Yes. Our IPDID and IP800 numbers support T.38
for faxing.
back to
top |
|
Does
IP Communications support Dynamic IP Addressing? |
Yes. With SIP Registration your IP Address can
change, and our systems will automatically be notified. Also,
we can route calls to domain names. So you can use Dynamic DNS
services like
www.dynip.com
or
www.dnsexit.com to map
dynamic IP to domain names.
back to
top |
|
How do I use your voip
service? |
IP Communications provides IP delivered toll
free and DID numbers worldwide. We can route these numbers to
your existing VoIP device or we can supply one for you.
Getting started with IP Communications is easy. Simply visit
our website at
www.ipcomms.net, and choose
the service and package you desire. Simply choose the phone
numbers or number locations you require and complete the order
process. Within 24 hours (usually less) of receiving your
order, you will receive a service provisioning letter that will
contain all the information you need to configure your VoIP device.
Configure your device with our information and begin placing and/or
receiving calls. You will be able to monitor and maintain your
service online using AMI our Account Management Interface found
here:
www.myipcomms.net.
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top |
|
I have my own box, but I am having
issues configuring it, can IP Communications help? |
Yes. When you sign up for IP
Communications services, our technical support team is available to
you and can assist with the configuration of many popular VoIP
devices.
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top |
|
My MC3810 VoIP router just rebooted and now my calls are not
working. |
Often when MC3810s reboot, the T1 channel does
not come back up by itself. You simply need to login to the
Router and bounce the T1 controller.
back to top |
|
What Gateway vendors
do you support? |
We support most Gateway vendors and protocols.
Our network has the ability to meet most connection requirements.
And yes, we do support Asterisk open source PBXs!
back to top |
|
Is IP
Communications compatible with Asterisk? |
Yes, we are compatible with the Asterisk
service.
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top |
|
What is Asterisk?
Is it really free? |
Asterisk is an open source software PBX,
created by Digium, Inc. and a continuously growing user and
developer base. Digium invests in both developing the Asterisk
source code and low cost telephony hardware that works with
Asterisk. Asterisk runs on Linux and other Unix platforms with OR
without hardware that connects your server to the traditional global
telephony network, the PSTN
Asterisk gives you real-time connectivity on both PSTN and VoIP
networks
With Asterisk as your telephony switching platform, PBX, you'll not
only have a high-class PBX replacement. Asterisk is much more than
the standard PBX. With Asterisk in your network, you can do
telephony in new ways.
-Connecting employees working from home to the office PBX over
broadband connections
-Connecting offices in various states over VoIP, Internet or a
private IP network
-Giving all employees voicemail, integrated with the Web and their
E-mail
-Building interactive voice applications, that connect to your
ordering system or other in-house applications
-Giving access to the company PBX for business travelers, connecting
over VPN from airport or hotel WLAN hotspots ...and much more
Asterisk includes many features only found in top-of-the-line
unified messaging systems, like
-Music-on-hold for customers waiting in queues, supporting streaming
media as well as MP3 music
-Call queues where call agents jointly handle answering incoming
calls and monitor the queue
-Text-to-speech system integration (the Festival Open Source and
Cepstral Swift speech synthesis software can be integrated)
-Call data record (CDR) generation for integration with billing
systems
-Voice recognition system integration (such as the Sphinx Open
Source voice recognition software)
-The ability to interface with normal telephone lines, ISDN basic
rate and primary rate interfaces
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|
Can all of my phone numbers be sent to the same IP Address? |
Yes. With most services we can
point your phone numbers to one IP address.
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|
Can I
port my number away from IP Communications |
Our IPDID and IP800 numbers are fully portable
both in and out.
back to
top |
|
How many concurrent calls can be placed on the numbers ? |
This depends on the capacity (number of
channels) that you have in your account. The more channels you have,
the more concurrent calls you will be able to receive.
back to
top |
|
Do you provide
real-time ANI / CLI? |
Yes. Our IPDID and US based IP800 numbers
provide ANI number (we do not support Caller ID name at this time).
back to
top |
|
What is Switchvox? |
The next generation of business phone systems
Find out more about Switchvox today!
Switchvox is everything that you don't expect from a PBX. It's truly
affordable, easy to set up, simple to configure, and a breeze to
maintain.
It has features that let your business run more effectively and with
fewer hassles. And it does all of this for a fraction of the cost of
the PBX dinosaurs of the past.
Switchvox is so much more than just an office phone system. Its a
revolution in business communications, putting you in control of
your most important asset in business, your voice.
With this incredible leap in technology comes astounding
cost-savings for your business, integration capability that you
never thought possible, and the flexibility to meet the needs of
whatever industry that you're in.
back
to top |
|
Do you have
a sample configuration for trixbox? |
|
4.1 What is FreePBX?
Asterisk
Management Portal makes Asterisk configuration easier by
providing a graphical method (through a web browser).
FREEPBX allow you con configure the textual
configuration files that Asterisk needs to function.
FREEPBX can
configure the following in asterisk:
Incoming Calls --- Specify where to
send calls coming from the outside
Extensions --- Add extensions and set
voicemail properties
Ring Groups --- Group extensions that
should ring simultaneously
Queues --- Place calls into queues and
allow them to be answered in order
Digital Receptionist --- Create voice
menus to greet callers
Trunks --- Set up trunks to connect to
the outside world
Outbound Routing --- Manage which
trunks outbound calls go out
DID Routes --- Specify the destination
for calls if their trunk supports direct inward dial
On Hold Music --- Upload MP3 files to
be played while users are on hold
System Recordings --- Record or upload
messages for specific extensions
Backup and Restore --- Create, back up,
and restore profiles of your system
General Settings --- Set basic dialing,
company directory, and fax settings
For IP
Communications configuration purposes we will need to
enable some of the modules in FreePBX
Please follow these steps:
- Open
your web browser and type
HTTP://YourAsteriskIPaddressHere
- Switch
to Admin Mode. (click on the switch link in the
upper right corner)
- Click
on the Asterisk Menu
- Select
Free PBX
- Click
on Tool
- Click
on Module Admin
- Enable
the following
-
Core
-
Voicemail
-
IVR
-
Ring Groups
-
Recording
-
Call Forward
-
Call Waiting
-
Do-Not-Disturb
-
Info Service
4.2 Configuring an extension
- Open
your web browser and type
HTTP://YourAsteriskIPaddressHere
- Switch
to Admin Mode. (click on the switch link in the
upper right corner)
- Click
on the Asterisk menu and select FreePBX.
- In the
FreePBX menu click setup and select extensions.
5. From the
device drop down menu select “Generic SIP device” and
click submit.
Example
- Create
extension 200 and type in a password for
registration like "abc123". Then enter the name of
the person using this extension.
- Select
enable, and enter a voicemail password. Use
something you can type on a phone keypad like
'1234'. Enter an e-mail address where you would like
your voice messages sent and click add extension.
Then click on the red apply bar at the top of the
screen.
-
Configure your extension in a soft phone for
testing. Xlite is the best choice for this test.
Remember to use your extension number and password
in Xlite. Use your Trixbox private IP address
as the sip proxy.
- Make a
call from your phone. Try *43. This is an echo test.
NOTE: If the
extension you are configuring will connect remotely
(outside the Local Area Network) you will need to change
the NAT option to yes.
Just create
the extension, submit the changes and go back to edit
it. You will see NAT=never; change it to NAT=yes
Every time you
make a configuration change and click “Submit” an ORANGE
button will appear at the top of the screen “Apply
Configuration Changes”. This button will reload the .
conf files. Click this bar in order for the changes to
take effect.
4.3 Configuring trunk for inbound calls
-
Connect to your Trixbox using a PC in your network
by typing
HTTP://YourAsteriskIpaddress
in your web browser.
- Select
FREEPBX under the “Asterisk” Menu
- Click
Trunks then “Add SIP Trunk”.
- Only
enter the following information:
Incoming Settings
User Context = sip.ipcomms.net
*Registration String = DO NOT ENTER REGISTRATION
STRING ON THIS SCREEN.
Leave this registration string text box empty. It
will be entered in the sip_nat.conf file.
- Click
“Submit Changes”
- Use a
pc on your network that has a web browser and
connect to your Trixbox box using
HTTP://PutYourTrixboxIpaddressHere.
If you have a public IP address,
nothing needs to be added to the general section go to
step 4.5.
4.4
What do I do if my asterisk server has a private IP
Address (Optional)?
- Click
on the Asterisk menu.
- Click
on Config Edit
- Click
on sip.conf
- Enter
the following information at the top of your
sip.conf file in the general section:
externip=WAN IP address
localnet=192.168.1.0/255.255.255.0 (or whatever your
private LAN is configured for)
- Click
UPDATE
-
Click
re-Read Configs located at the top of the screen.
Note:
If public, nothing needs to be added to the general
section. Depending on your setup, you might need
to forward all ports to your Asterisk server's private
IP address or put the private IP address of the Asterisk
server on the DMZ.
4.5
Configuring Inbound Routes
NOTE: YOU WILL
NOT BE ABLE TO RECEIVE CALLS IF YOU DO NOT CONFIGURE AT
LEAST ONE INBOUND ROUTE
Configuring
inbound routes will allow calls from IP Comms to go
someplace in your PBX.
Using
FREEPBX
- Select
setup
- Select
Inbound Routes.
- Leave
the DID number and Caller ID Number boxes
empty.
- Under
“set destination” select extension 200.
- Click
Submit
Call the your IP Communications DID . Your SIP phone
extension should ring.
|
back to top |
|
How do you make a T1
Crossover cable? |
If you have some CAT5 wire, a pair of new RJ45 ends, and a crimping
tool, you can make your own t1 crossover cable. This cable is useful
if you will be connecting two systems together with a T1 circuit in
the same room.
From looking at the picture below, the
clips of the RJ-45 plug should be facing down. Put plug1 on one side
of the cat5e cable, plug2 on the other and crimp.
For video instruction on creating a network cable,
click here
(*This video demonstrates a
straight-through configuration, for crossover, use the
pin-out below),
Use this pin-out:
|
|
How does SIP carry DTMF? |
There are at least two options for carrying DTMF and similar signals
in a VoIP network using SIP. First, DTMF can be transported as an
RTP payload (RFC 2833). This has the advantage that it provides
accurate timing and alignment with the speech RTP packets. Also,
media gateways are the most likely to detect and generate tones, so
that making it part of the media stream is appropriate. However,
under some circumstances, it may be necessary for signaling entities
to know about DTMF signals. Currently, there is no standardized
solution within SIP, but it has been proposed to carry DTMF
information in SIP INFO messages, either encoded as simple text or
using the RFC 2833 format. The latter is more complex, but offers
duration and timing information.
back
to top
|
|
What is IVR /
interactive voice response? |
Interactive Voice Response or IVR is a telephone technology that
communicates with a caller through configurable voice menus and data
in real time. In an IVR system, callers are given the choice to
select options by pressing digits.
IVR systems can normally handle and service high volumes of phone
calls. With an Interactive Voice Response system, businesses can
reduce costs and improve customers’ experience as Interactive Voice
Response systems allow callers to get information they need 24 hours
a day without the need of costly human agents.
Some IVR applications include telephone banking, flight-scheduling
information and tele-voting.
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|
What is an auto-attendant? |
Auto-attendant (or automated attendant) is a term commonly used in
telephony to describe a voice menu system that allows callers to be
transferred to an extension without going through a telephone
operator or receptionist. The auto-attendant is also known as a
digital receptionist.
For a caller to find a user on a phone system, a dial-by-name
directory is usually available. This feature lists users by name,
allowing the caller to press a key to automatically ring the
extension of a user once his/her extension is announced by the auto
attendant.
If a user is not available, the auto-attendant directs callers to
the appropriate voice mailbox of the user to leave a voicemail
message.
Having an auto-attendant in a phone system is a very useful and
cost-effective feature for a business, as it replaces/helps the
human operator by automating and simplifying the incoming phone
calls procedure.
back to
top |
| Can you provide a complete list of colocation/data center locations?
|
Market |
Address |
City |
State |
Zip Code |
Milwaukee |
324 E Wisconsin |
Milwaukee |
WI |
53202 |
Columbus |
240 N 5th Street |
Columbus |
OH |
43215 |
Pittsburgh |
650 Smithfield St. |
Pittsburgh |
PA |
15222 |
Boca Raton |
5050 Conference Way North |
Boca Raton |
FL |
33431 |
New York |
33 Whitehall |
New York |
NY |
10004 |
Atlanta |
55 Marietta St NW |
Atlanta |
GA |
30303 |
Wash, DC |
1120 Vermont Ave NW |
Washington |
DC |
20005 |
Oakland |
1624 Franklin Street |
Oakland |
CA |
94612 |
Boston |
59 Innerbelt Road |
Somerville |
MA |
2143 |
Chicago |
427 S LaSalle |
Chicago |
IL |
60605 |
Chicago |
600 S Federal St |
Chicago |
IL |
60605 |
Cleveland |
1621 Euclid Ave |
Cleveland |
OH |
44115 |
Los Angeles |
530 W 6th St |
Los Angeles |
CA |
90014 |
Wash, DC |
1921 Gallows Rd |
Vienna |
VA |
22182 |
Wash, DC |
510 Huntmar (Hosting Ctr) |
Herndon |
VA |
20170 |
Chicago |
216 W Jackson Blvd |
Chicago |
IL |
60606 |
Dallas |
400 S Akard St |
Dallas |
TX |
75202 |
Miami |
200 SE 1st St |
Miami |
FL |
33131 |
Orange County |
300 S Harbor Blvd |
Anaheim |
CA |
92805 |
back to
top
|
SIP responses are the codes used by Session Initiation Protocol for communication. They complement the SIP Requests, which are used to initiate action such as a phone conversation. Note that the Reason Phrases of the responses listed below are only the recommended examples, and can be replaced with local equivalents without affecting the protocol.
1xx—Informational Responses
extended search being performed may take a significant time so a forking proxy must send a 100 Trying response
- 180 Ringing
- 181 Call Is Being Forwarded
- 182 Queued
- 183 Session Progress
2xx—Successful Responses
- 200 OK
- 202 accepted: It Indicates that the request has been understood but actually can't be processed
3xx—Redirection Responses
- 300 Multiple Choices
- 301 Moved Permanently
- 302 Moved Temporarily
- 305 Use Proxy
- 380 Alternative Service
4xx—Client Failure Responses
- 400 Bad Request
- 401 Unauthorized (Used only by registrars or user agents. Proxies should use proxy authorization 407)
- 402 Payment Required (Reserved for future use)
- 403 Forbidden
- 404 Not Found (User not found)
- 405 Method Not Allowed
- 406 Not Acceptable
- 407 Proxy Authentication Required
- 408 Request Timeout (Couldn't find the user in time)
- 409 Conflict
- 410 Gone (The user existed once, but is not available here any more.)
- 412 Conditional Request Failed
- 413 Request Entity Too Large
- 414 Request-URI Too Long
- 415 Unsupported Media Type
- 416 Unsupported URI Scheme
- 417 Unknown Resource-Priority
- 420 Bad Extension (Bad SIP Protocol Extension used, not understood by the server)
- 421 Extension Required
- 422 Session Interval Too Small
- 423 Interval Too Brief
- 424 Bad Location Information
- 428 Use Identity Header
- 429 Provide Referrer Identity
- 433 Anonymity Disallowed
- 436 Bad Identity-Info
- 437 Unsupported Certificate
- 438 Invalid Identity Header
- 480 Temporarily Unavailable
- 481 Call/Transaction Does Not Exist
- 482 Loop Detected
- 483 Too Many Hops
- 484 Address Incomplete
- 485 Ambiguous
- 486 Busy Here
- 487 Request Terminated
- 488 Not Acceptable Here
- 489 Bad Event
- 491 Request Pending
- 493 Undecipherable (Could not decrypt S/MIME body part)
- 494 Security Agreement Required
5xx—Server Failure Responses
- 500 Server Internal Error
- 501 Not Implemented: The SIP request method is not implemented here
- 502 Bad Gateway
- 503 Service Unavailable
- 504 Server Time-out
- 505 Version Not Supported: The server does not support this version of the SIP protocol
- 513 Message Too Large
- 580 Precondition Failure
6xx—Global Failure Responses
- 600 Busy Everywhere
- 603 Decline
- 604 Does Not Exist Anywhere
- 606 Not Acceptable
|
|
Local Exchange Service |
Local exchange service
provides calling within your exchange. An exchange is a
specified area which usually encompasses a city, town,
or village and its environs. |
|
Local Toll (intraLATA)
Service |
Local toll service (also
called intraLATA, local long distance, or regional toll
service) provides calling within a geographic area known
as a Local Access and Transport Area (LATA). Per-minute
toll charges usually apply to these calls. LATAs were
formed in 1984 when the former Bell System was broken up
into Bell Operating Companies, which handled local
exchange and local toll services, and AT&T, which
handled interLATA long distance service in competition
with other long distance providers. Today, these
companies (or the companies they have become) handle all
types of calls, but LATAs still define local toll
calling areas.
Local toll calls may be made within your area code or to
a different area code across town, in the next county,
or, in some cases, an adjoining state. You must dial “1”
before making a local toll call, even if the area code
is the same as yours. Today these calls can be carried
by your local exchange telephone company or your long
distance company.
Some local telephone companies offer an optional bundle
of local exchange and local toll service for a single
monthly fee.
|
|
Expanded Local Exchange
Service |
Expanded local exchange
service extends a local exchange calling area and
eliminates local toll costs; however, you may see
expanded local exchange service as a surcharge on your
telephone bill. |
|
Long Distance Toll
(interLATA) Service |
Long distance toll
(interLATA) service includes all calls outside the local
exchange and local toll service areas, calls that
originate in one LATA and terminate in another, and
international calls. Long distance toll calls can be
between two LATAs in the same state, such as a call from
San Diego to San Francisco, or between LATAs in
different states. Long distance toll service includes
international service, except in Hawaii, where
international service is separate from long distance
service. When purchasing long distance toll service,
remember to specifically ask whether international calls
are included in monthly long distance calling plans. |
|
PSTN (Public Switched
Telephone Network) |
The PSTN is the means by
which most landline telephone calls are routed. It is
the collection of interconnected systems operated by the
various telephone companies and administrations around
the world. The PSTN is also known as the Plain Old
Telephone System (POTS). |
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Cellular Network |
Cellular Network |
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SIP |
SIP stands for Session
Initiation Protocol and is a set of rules that govern
the format of the coded messages which are exchanged
during an Internet call. SIP is the most popular VoIP
standard because it is an open standard.
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Codec |
Codec is a term that arises
from the Compressor-Decompressor or enCOder/DECoder
process. It is used for software or hardware devices
that can convert or transform a data stream. For
instance, at the transmitting end codecs can encode a
data stream or data signal for easy transmission,
storage or encryption. At the receiving end, they can
decode the signal in the appropriate form for viewing.
They are most suitable for videoconferencing and
streaming media solutions. |
|
Analog audio signals |
Analog
audio signals are used to transmit voice data over
telephone lines. This is done by varying or modulating
the frequency of sound waves to accurately reflect the
pitch of the sound. The same technology is used for
radio wave transmissions. |
|
ATA |
ATA
or the analog telephone adaptor is the hardware device
that connects the conventional telephone to the Internet
through a high speed bandwidth line, provides the
interface to convert the analog voice signals into IP
packets, delivers dial tone and manages the call setup.
More on ATAs... |
|
Bandwidth |
Bandwidth
is the volume of data that can be transmitted over a
communication line in a fixed amount of time. It is
expressed in bits per second (bps) or bytes per second
for digital devices and in cycles per second, or Hertz
(Hz) for analog devices. Bandwidth can also be defined
as the difference between a band of frequencies or
wavelengths. |
|
Broadband |
It
is a term used to define high speed Internet connection,
generally provided by cable TV, DSL or dedicated telecom
lines. The high speeds are achieved by the carrying
capacity of the cable that can carry multiple messages
simultaneously. |
|
Cable modem |
The
cable modem is a device that is used to connect a
computer to the high speed coaxial cable run by cable TV
companies to provide access to the Internet. The
connection is made through an Ethernet port, which is a
shared medium and can affect download speeds if too many
users log on simultaneously to the Internet on that
particular cable segment. However, despite this cable
modems provide extremely fast access to the net. |
|
Circuit switched networks |
These
networks have been used for making phone calls since
1878. They use a dedicated point-to-point connection for
each call. This reduces their utility because no network
traffic can move across the switches that are being used
to transmit a call. |
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Client (Softphone client) |
The
software installed in the user’s computer to make calls
over the Internet. More on softphones... |
|
Codec |
Codec
is a term that arises from the Compressor-Decompressor
or enCOder/DECoder process. It is used for software or
hardware devices that can convert or transform a data
stream. For instance, at the transmitting end codecs can
encode a data stream or data signal for easy
transmission, storage or encryption. At the receiving
end, they can decode the signal in the appropriate form
for viewing. They are most suitable for
videoconferencing and streaming media solutions.
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Compression |
This
is a term that is used to indicate the squeezing of data
in a format that takes less space to store or less
bandwidth to transmit. It is very useful in handling
large graphics, audio and video files. |
|
Data compression |
This
is the process that is used to compress large data files
into mall files so that they use less bandwidth during
transmission and less disk space when stored. The
compression depends upon the repeatable patterns of
binary 0s and 1s. The higher the number of repeatable
patters, the higher is the compression. The right
compression codes can compress data files to 40% of
their original size. The graphics files can be
compressed even more – from 20% to 90%. |
|
DID |
Short for direct inward
dialing (also known as direct dialing inward), a service
of an LEC or local phone company that allows an
organization to have numerous individual phone numbers
for each person or workstation in its PBX system that
run off of a small block of dedicated telephone numbers.
DID allows the multiple lines to be connected to the PBX
all at once without requiring each to have a physical
line connecting to the PBX.
For example, if an organization has 25 employees and
each employee has a separate telephone number, or
extension, within its physical location, the
organization can rent 10 physical trunk lines from the
telephone company that will allow 10 phone calls to take
place simultaneously. Others would have to wait for an
available line and anyone dialing into the system while
all 10 lines are in use would get either a busy signal
or be channeled into a voice mail system. A DID system
does not require a PBX operator and can be used for fax
and voice transmissions.
|
|
DSL modem |
A
DSL modem is a device that is used to connect one or
more computers to the high speed DSL line provided by a
DSL operator to gain access to the Internet. The
customers use these modems to log on the net to download
or transmit data. Since the DSL lines have high
bandwidth capacity the data transfer speeds are very
high. |
|
E911 |
E911
is the short form of the term Enhanced 911, and is used
for providing emergency service on cellular and Internet
voice calls. |
|
Emergency 911 calls |
This
is an emergency telephone number that handles all calls
related to police, fire or medical emergencies. The
number, which is allotted under the North American
Numbering Plan (NANP), is answered by either a telephone
operator or an emergency service dispatcher, who, in
turn, alerts the appropriate emergency service. |
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H.323 |
An
ITU standard that lays down guidelines for real time
voice and videoconferencing utilities on the Internet.
The H.323 standard supports voice, video, data,
application sharing and whiteboarding and defines media
gateways for conversion to packets. |
|
IM |
IM,
which stands for Instant Messenging, is a software that
allows users to exchange messages in real time. However,
to do so both the users must be logged on to the instant
messaging service at the same time. Some of the popular
IM services are |
|
Internet congestion |
Internet
congestion occurs when a large volume of data is being
routed on low bandwidth lines or across networks that
have high latency and cannot handle large volumes. The
result is slowing down of packet movement, packet loss
and drop in service quality.
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|
IP |
IP,
which is the acronym for Internet Protocol, defines the
way data packets, also called datagrams, should be moved
between the destination and the source. More
technically, it can be defined as the network layer
protocol in the TCP/IP communications protocol suite. |
|
IP address |
An
IP address, also known as Internet Protocol address, is
the machine number used to identify all devices that are
connected to the net. Each device has its own unique
number which it uses to communicate. This number is
fixed in the case of those computing devices that have a
fixed IP address. The rest are allotted a dynamic IP
address, which is valid for the period they are
connected to the net. The numbers range from 0.0.0.0 to
255.255.255.255. |
|
IP mapping |
IP
mapping is the process of identifying IP addresses on
the basis of their geographical locations. The mapping
enables web administrators to pinpoint the location of
any computing device connected to the Internet. |
|
IP Phone |
An
IP phone is one that converts voice into digital packets
and vice versa to make phone calls over Internet
possible. It has built-in IP signaling protocols such as
H.323 that ensure that the voice is routed to the right
destination over the net. The IP phones come with
several value added services like voicemail, e-mail,
call number blocking etc. |
|
IP telephony |
IP
telephony refers to the two-way transmission of voice
over Internet. The voice is transmitted in real time by
using the packet-switched technology over the IP
network. Some of the applications that use IP telephony
are IP-based phone services, voice over instant
messaging and videoconferencing. |
|
ITU |
ITU,
which is the acronym of International Telecommunication
Union, is a telecommunications standards body based in
Geneva. It works under the aegis of the United Nations
and makes recommendations on standards in
telecommunications, information technology, consumer
electronics, broadcasting and multimedia communications. |
|
Jitter |
It
is a term used to indicate a momentary fluctuation in
the transmission signal. This happens in computing when
a data packet arrives either ahead or behind a standard
clock cycle. In telecommunication, it may result from an
abrupt variation in signal characteristics, such as the
interval between successive pulses. |
|
Kbps |
Kbps
is the acronym for kilobits per second and is used to
indicate the data transfer speed. If the modem speed,
for instance, is 1 Kbps then it means that the modem can
route data at the speed of one thousand bits per second. |
|
Lag |
Lag
is the term used to indicate the extra time taken by a
packet of data to travel from the source computer to the
destination computer and back again. The lag may be
caused by poor networking or by inefficient or excessive
processing. |
|
Latency |
Latency
is the time that elapses between the initiation of a
request for data and the start of the actual data
transfer. This delay may be in nanoseconds but it is
still used to judge the efficiency of networks. |
|
Mapping |
The
process of identifying all related data fields or data
streams and putting them in an easily identifiable
context. For example, IP mapping enables users to
pinpoint the geographical location of any computing
device on the Internet. |
|
MGCP |
Acronym
of Media Gateway Control Protocol. Used for a Voice over
IP system. It consists of a Call Agent and a set of
gateways, of which at least one works as the "media
gateway" and performs the conversions. |
|
NANP |
Stands
for North American Numbering Plan. A telephone numbering
system that has evolved the way area codes and numbers
are allotted. The system was established in 1947 and
covers the United States, Canada and a few neighboring
areas. It uses a three-digit area code and seven-digit
telephone numbers. Its fiat is, however, limited to the
public switched telephone networks only. |
|
Net Phone |
A
net phone uses the Voice over IP technology to make
voice calls. These calls are made by converting analog
sound signals into digital data packets, and then moving
the packets to their destination over the net.
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Packet |
A
packet is a unit of data transmitted over the network in
a packet-switched system. It consists of a header that
stores the destination address, a data area which
carries the information that is being transmitted, and a
trailer which contains information to prevent errors
during transmission. |
|
Packet loss |
Packet
loss is the term used to indicate the loss of data
packets during transmission over a computer network.
This may happen on account of high network latency or on
account of overloading of switches or routers that are
unable to process or route all the incoming data. |
|
Packet switched networks |
These
are networks that break messages into small digital
packets, stamp each packet with the destination IP
address, and route them across different channels to
their destination where they are reassembled in their
proper sequence. This is done to avoid network
congestion and speed up data movement from multiple
sources. |
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Peer-to-Peer (P2P) |
The
term peer-to-peer is used to indicate a form of
computing where two or more than two users can share
files or CPU power. They can even transmit real time
data such as telephony traffic on their highly ad hoc
networks. Interestingly, the peer-to-peer network does
not work on the traditional client-server model but on
equal peer nodes that work both as "clients" and
"servers" to other nodes on the network. |
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POTS |
POTS
is the short form of plain old telephone service. It
transmits voice as analog data on communication lines
that are much slower when compared to today’s ISDN or
FDDI lines. However, not long ago POTS, which is also
known as the public switched telephone network, was the
standard telephone system across the world. |
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Processor drain |
This
is a term used to indicate a drop in the quality of VoIP
phone service when a user opens several applications on
his computer simultaneously. |
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Protocol |
It
is a convention or standard that defines the procedures
to be adopted regarding the transmission of data between
two computing end points. These procedures include the
way the sending device should sign off a message or how
the receiving device should indicate the receipt of a
message. Similarly, the protocols also lay down
guidelines for error checking, data compression, and
other relevant operational details. |
|
PSTN |
PSTN,
which stands for Public Switched Telephone Network,
refers to the telephone system that transmits analog
voice data. Till recently, PSTN was the heart of all
phone systems worldwide. However, most of the developed
world is now switching to or has switched to telephone
networks that are based on digital technologies, such as
ISDN and FDDI. RJ45 |
|
Router |
A
router is a network device that that handles message
transfer between computers that form part of the
Internet. The messages, which are in the form of data
packets, are forwarded to their respective IP
destinations by the router. A router can also be called
the junction box that routes data packets between
computer networks.
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|
Sampling |
This
is a methodology used to measure the value of an analog
signal at regular intervals, and encoding it into a
digital format for VoIP phone services. |
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Service provider |
A
service provider is a business entity that provides a
communication, storage or processing service for a fee.
Some of the service providers in the digital world are
the Internet service provider (ISP), application service
provider (ASP), storage service provider, mobile phone
service provider, web hosting provider, and of course,
VOIP service provider. |
|
SIP |
SIP,
which is the acronym of Session Initiation Protocol, is
an IP telephony signaling protocol. It is primarily used
for voice over IP (VoIP) calls, though with some
extensions it can also be used for instant messaging. It
is less complex than H.323, the other IP telephony
protocol. |
|
SIP phone |
A
SIP phone is a telephone that uses the SIP (Session
Initiation Protocol) standard to make a voice call over
the Internet. The SIP phones come with several value
added services like voicemail, e-mail, call number
blocking etc. There are no charges for making calls from
one SIP phone to another, and negligible charges for
routing the call from a SIP phone to a PSTN phone. |
|
Skype |
Skype
is a peer-to-peer Internet telephony company that
revolutionized the way voice calls are made by using
VoIP technology. The company, which has been acquired by
eBay, was founded by Niklas Zennstr? and Janus Friis.
Skype users can speak to other Skype users for free, but
have to pay a small fee for calling or receiving calls
from conventional phones. |
|
Soft switch |
It
is a software application that is used to keep track of,
monitor or regulate connections at the junction point
between circuit and packet networks. This software is
loaded in computers and is now replacing hardware
switches on most telecom networks. |
|
Softphone |
This
is a software application that is installed in the
user’s PC. It uses the Voice over IP technology to route
voice calls over the net and provides several value
added features, such as call forwarding, conference
calling, and integration with applications such as
Outlook for automatic dialing The audio is provided
through a microphone and speakers plugged into the sound
card. The only limitation of a Soft phone is that the
phone call has to made through a PC. Many soft phone are
free VOIP software downloads. |
|
Term |
Definition |
|
Voice chat |
This
is an application that enables two or more than two
individuals to carry on a verbal conversation over the
Internet. Voice chat is also known as audio-conferencing
or telephone conferencing on the net. |
|
Voice over IP (VOIP) |
VoIP
or Voice over IP is the technology that is used to
transmit voice over the Internet. The voice is first
converted into digital data which is then organized into
small packets. These packets are stamped with the
destination IP address and routed over the Internet. At
the receiving end the digital data is reconverted into
voice and fed into the user’s phone.
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Voicemail |
It
is a telephone messaging system that digitizes the
analog voice signals and stores them on disk or flash
memory in a central computer. These messages can then be
retrieved by users by logging on to the server or
forwarded to another voice mailbox. Most voice mail
systems have auto attendant capabilities, that is they
can use prerecorded messages to route callers to the
appropriate person or mailbox. Voicemail is usually a
free feature in VOIP service plans |
|
VOIP Gateway |
This
device provides the conversion interface between the
public switched telephone network (PSTN) and an IP
network for voice and fax calls. Its primary functions
include |
|
VOIP PBX |
VoIP
PBX, which stands for Voice over Internet Protocol
Private Branch eXchange, is a telephone switch that
converts IP phone calls into traditional
circuit-switched TDM connections. It also supports
traditional analog and digital telephones. |
|
VOIP Phone |
A
VoIP phone is one that uses the Internet to route voice
calls by converting the voice data into IP packets and
vice versa. The phones come with built-in IP signaling
protocols such as H.323 or SIP that help in the routing
of data to the right destination. A VoIP phone can also
be a software application that is installed in the
user's PC. In this case it is known as the Softphone.
Also, the calls in this case have to be made from the
PC, and not through a telephone instrument. |
|
VOIP services |
The
VoIP services are packet-based services that use the
Internet to move voice data. These services are much
cheaper than the traditional PSTN services because the
investment in infrastructure is low. They also come with
several value added features which make them more
lucrative than the conventional landline phone services. |
|
Web phone |
A
web phone is a device that allows users to make voice
calls over the Internet. |
|
WiFi Hotspot |
An
area where a wireless access point enables users
carrying wireless-enabled laptops to log on to the
Internet. The limiting condition is that the access
point is configured to broadcast its presence and does
not require authorization for access. Generally, WiFI
hotspots are located in public places like airports,
train stations, libraries, marinas, convention centers,
coffee shops and hotels. |
|
WiFi phone |
A
WiFI phone is one that enables users to make phone calls
from public WiFi hotspots or residential WiFI network
environments. Besides voice calls, these phones can be
used to send e-mails wirelessly.
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