Asterisk PBX

Asterisk PBX
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Asterisk PBX
Executive Summary
Asterisk is an open source platform for converged telecommunications. It provides
PBX functions and applications, as well as connectivity via TDM and packet voice.
Asterisk uses host processing for TDM and DSP, a lightweight protocol (IAX) for packet
voice, and a flexible application-centric architecture for PBX services. It also provides
interoperability with other VoIP protocols such as SIP, MGCP, and H.323. These features
plus a modular, flexible, and expandable design, enable enterprises as well as individuals
to deploy VoIP services in a very cost-effective manner. Asterisk is supported by Digium,
IAXTEL network, and the open source community.
Introduction
Most of the existing Asterisk literature almost exclusively provides technical
details focused on its configuration, setup, and integration. This article explains why
Asterisk is important and innovative without going into such technical details. It helps
executives and managers in converged telecommunications market have a better
understanding of key aspects of Asterisk as a VoIP platform. It also shows how an open
source project can be used as a cornerstone in a business model. Asterisk is an open
source converged telecommunications platform, designed to allow different types of IP
telephony hardware, middle ware, and software to interface with each other
consistently. It provides multiple layers, managing both TDM and packet voice at lower
layers while offering a highly flexible platform for PBX and telephony applications such
as IVR. Asterisk can bridge and translate different types of VoIP protocols like SIP, MGCP,
and H.323. At the same time it can provide a full-featured server platform for predictive
dialing, custom IVR, remote and central office PBX, and conferencing. The name Asterisk
refers to the “*” symbol which is a “wild card” in Unix and DOS command line syntax,
denoting a symbol of a very versatile component in a voice network. Implementing host
based DTM and DSP, allowing multiple packet voice protocols to interact, and offering a
modular design with APIs for adding new applications have made Asterisk a real wild
card in converged telecommunications.
The existing phone service in many homes is analog Plain Old Telephone Service
POTS for incoming lines from the phone company, also known as the Public Switched
Telephone Network (PSTN) connected to regular analog phones. (Note: In the US ISDN
(Integrated Services Digital Network), digital phone lines, never really took off the way
in did in some parts of Europe and Asia.) This page covers analog phones and phone lines
only, in particularly the low end devices that are most appropriate for home use.
Asterisk can be used in all digital configurations using Voice over IP (VoIP)
connections to software phones running on PC and to phone service providers that you
can connect to over the Internet. There are a lot of tutorials explaining how to do that.
Those can be quite helpful for learning and getting started. However, once you want to
start using Asterisk "for real", you'll probably want to integrate asterisk with your
existing home phones and phone lines. I found a lot of the information was scattered in
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different places, so this page is an Attempt to help people get started and help make an
informed decision on what hardware to get.
While there is an ever growing set of choices for digital VoIP phones, there is a
much larger selection of plain analog phones available practically everywhere at very
low price points. In terms of the economies of scale, look at how much smaller the
choices are for 2 line phones. The handful of phones that are available with more than 2
lines that I've seen can cost about as much as your whole Asterisk installation. The
majority of the available VoIP/SIP phones are hardwired. I believe the most desirable
phones for home are cordless. Things are improving however, two of the largest
consumer cordless phone producers, Uniden and V Tech have announced cordless VoIP
phones. Hybrid mobile phones that can do VoIP over WiFi when it's available and fall
back to a cellular network (GSM, CDMA) look very promising, but it's still pretty early. I
believe most high tech households that would be considering installing Asterisk probably
already have a reasonable investment in the better cordless phones.
If your home needs a lot of phones, implementing Asterisk could allow you to save
quite a bit of money allowing you to buy fairly simple low end phones. If you are reading
this, you probably already know this but there are many reasons for introducing the
Asterisk PBX into your existing home phone setup such as:
Sophisticated Voice Mail system to replace that aging answering machine. This can
provide a mail box per person, that can be deliver notification by e-mail. Web
based access to your voice mail is also available.
Interactive Voice Response IVR systems - You can present callers with a menu,
which can be particularly useful if you have more people in the house than you
have incoming phone lines. "Press 1 for Him, Press 2 for Her, Press 3 for Kid No. 1,
Press 4 for Kid No. 2"...
Control over which phones ring at what times.
Intercom (Place in house calls)
Routing incoming calls by Caller ID.
Need more than 2 incoming lines (Phones that handle more than two lines are
much more expensive than 1 or 2 line phones, and there isn't very much selection
available.) Call Detail Reports (for attempting to gain some control over costs,
and/or teenagers)
Benefits
Benefit (Voice Network)
PURE IP-PBX System
Working on TCP/IP protocol
Working on Wireless/LAN/WAN network
■ Roaming Extensions
■ Branch Office on WAN
Convergence - Voice/Video/Data Integration on single connection
High Availability Service (HA)
IAX protocol
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Cascading (Dial-plan)
Benefit (Cost Saving)
Low implementation cost (*depend on system complexity)
IVR server
Call Center
Cheap calling rate (VoIP)
Free Inter-branch, remote extension call (IAX)
Lower International call cost (Direct VoIP Call)
Cost-effective migration path
Don’t have to replace whole system
Can connect to legacy system (various interface card supported)
Benefit (Open Standard Protocols)
By using SIP/H.323 protocols, any SIP compliance devices - phone, Adapter, Soft
phone, WiFi phone can be used.* (*some feature are not available on some phone)
Compatible with many public SIP servers (VoIP Operator)
Thailand
■ JINET
■ CAT
■ TRUE
■ Etc...
International
■ VoIP stun
■ Gizmo's
■ Etc…
Benefit (Add/Change/Remove)
Easy to Add/Change/Remove Extensions.
Re-route at patch panel not require
IP requires network connection
Reduce technical support for phone system (only IT STAFF required)
Movable to any location under the same LAN/Networking infrastructure.
Benefit (Custom Features)
New features can be added (by administrator)
Wake-up call
Phone Directory
Web – Based / Email Voice mail
New Application can be easily implemented
Personal Voice Menu
Intelligence Home System
Voice-based Control System
Many more…
Benefit (Standard Features / Software)
Support most advanced features found in the enterprise-grade PABX system
Support 3rd party software
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CRM (Sugar CRM)
Billing System
FOP (Flash Operator Panel)
Etc...
Ready to use with bundled software (TrixBox)
Benefit (support)
Large community on the Internet.
Growing community in Thailand.
Training sessions available through many
Universities in Thailand
Asterisk Features
Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk
offers both classical PBX functionality and advanced features, and inter operates with
traditional standards-based telephony systems and Voice over IP systems. Asterisk offers
the advanced features that are often associated with large, high end (and high cost)
proprietary PBXs.
Auto-Attendant
Auto Attendant serves as an automated receptionist that answers calls and
provides a personalized massage to callers with menu options to direct their calls. For
example Press 1 for Sales or press 2 for support, etc. Auto-Attendant is simple to
operate and powerful. By simply clicking on mouse you can schedule, forward, respond,
manage of call flow. It also has following features
Dial by Name: Auto attendant allows in-bound callers to route their call to the
appropriate person without knowing their extension. This feature allows for either first
or last name directory look up.
Dial by Extension: In-bound callers may quickly route their call to the appropriate
person if they know the correct extension number.
Dial by Group: Will allow in-bound callers to route their call directly to the auto
attendant of a group or department.
Unlimited Extensions
With just few clicks of mouse you can create unlimited extensions and every
extension can be customized with unlimited options like DID (direct in dialing) privacy
setting, name directory by spelling, outbound caller ID, voice mail settings, etc.
Interactive Voice Response (IVR)
Asterisk's flexible IVR capability allows a user to interact with a database using a menu
of pre-recorded voice-clips. Using MySQL and other popular databases, Asterisk can
interact with the caller through touch tone inputs, record responses, query databases,
and
utilize
AGI
scripts
to
perform
specific
tasks.
For example, a customer can authenticate a pre-paid calling card with a PIN queried
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from a database. The Asterisk IVR will give the number of remaining minutes and later
disconnect if that customer runs out of time. The spooling feature can allow Asterisk to
dial a list of numbers from a database to give warnings during homeland security
emergencies. Asterisk IVR allows developers to create a myriad of IVR solutions.
Voice Mail
This feature enables users to record messages for incoming calls that are not
answered within a specified number of rings, or receive busy treatment, or are
transferred directly to voice mail. Asterisk comes with voice mail storage with storage
capacity of more than 1,000 hours and it can be retrieved from your office phone or you
can access by dialing from any remote phone or it can be attached with your email
as .WAV file or you can listen thru web control panel or to the voice messaging system
repository for retrieval from a phone. By accessing Voice Portal from any phone, user
can listen, save, delete, pausing, skip forward or back or reply to message or can be
forwarded to one or more or to entire group member with introductory message. Users
have the option of marking a message as Urgent or Confidential.
Voice mail to Email
You can receive your voice mail as an email via .wav file attached to your email
with a few click of mouse without configuration POP3, SMTP, IMAP or Exchange. If
available, the caller’s name and number are also included in the e-mail subject line..
Voice mail may be forwarded to email on an ad-hoc or continuous basis, giving the user
an easy-to-manage unified messaging system.
Music-on-Hold
Our product comes with multiple audio channels and can play different audio to
different class of caller on hold. By uploading an MP3 file using web control panel, your
caller will hear your favorite tune while on hold. Administrators can upload an audio file
(.wav file containing music, advertising, promo, etc.) onto the system to be broadcast to
held parties. This service can also be used in conjunction with the other services like
Call Hold, and Call Park.
Paging / Parking
This feature supports system-wide paging and single phone intercom (check the
phone models supporting for paging) or unlimited parking of calls simultaneously.
Call Parking - Enables a user to hold a call and to retrieve it from another station within
the group. To park a call, a user presses the flash hook and dials the call park feature
code. The call is parked and the caller is held. To retrieve the call, the user goes to any
phone in the group and dials the call retrieve feature code, followed by the user’s
extension. The call is retrieved and connected to the retrieving user.
Loudspeaker Paging - Enables users to access an intercom paging system by dialing an
extension within the group.
Call Conference
The Call Conference is an Asterisk solution-based PBX system. The Call Conference
provides conference room system for use by all users. The Call Conference from NSSF
PBX have any of the following features: Security passwords to control access to who can
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call into a conference bridge, Conference Leader pass codes that restrict the callers
from talking to one another until the conference leader has logged in, Name Announce
which asks the caller to announce themselves and then plays their name when they call
prior to placing the caller into the bridges
Our product comes with pre-configured bridge to support unlimited internal as
well as external participants as per your phone lines and if you are using our network
you can have virtually number of external participants. Conferencing - Web +
Audio/Video conferencing, users have an on-demand conference bridge and the ability
to schedule conference calls or web conferencing via the Personal Communications
Manager Web tool, eliminating 3rd party conference vendors.
Three-way Calling - Enables a user to make a three-way call with two parties, in which
all parties can communicate with each other. To initiate a three-way call while engaged
in a regular two-party call, the user presses the flash hook and dials the third party.
Before or after the third party answers, the user presses the flash hook and forms a
three-way call with the two parties. To drop the third party, the user presses the flash
hook and is reconnected with the original party in a regular two-party call. If the user
hangs up, all parties are released.
Analog & IP Phones
The Asterisk PBX supports all analog and various IP phones. It also supports
message waiting indicator on IP phone and on analog phone uses stutter-tone to indicate
voice mail waiting. Analog phones support multiple call appearances via call-waiting
flash-hook
Ring-all (Blast Group)
With this feature, one inbound call and it rings all phones, of course the first one
to pick gets the call. You can also configure which phones goes to blast group.
Simultaneous Ring (Personal)- Simultaneous Ring enables users to have multiple phones
ring simultaneously when any calls are received on their phone number, e.g. calls to a
user’s desk phone could also ring the user’s mobile phone, in case the user is not at his
desk. The first phone to be answered is connected.
Call Forwarding
Users can use their personal Web Control Panel to enable call forwarding to either
an internal extension or to an external number.
Call Forwarding Selective - Automatically forward your incoming calls to a different
phone number with pre-defined criteria, such as the phone number, time of day or day
of week, are met.
Name Directory
Once pre-configured by spell-by-last-name one can simply spell first three letters
of the party’s last name and it automatically connects to requested extension.
Custom Caller-IDs
You can customize outbound caller-ID, block, reveal or change caller-ID of every
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extension in office
Calling Group ID
Provide the name and number of the group (or company) for outgoing calls from
users in the group, rather than providing the user’s own name and number.
DIDs
When you have more employees than your phone number you can simply assign
any incoming call to an employee with just one click of mouse and it’s that easy..
VoIP-ready
With our VoIP enabled network you can save on long distance and international
calls. We can provide you low cost and high quality VoIP service. Opt for our VoIP interoffice and intra-office or telecommuters to make communication totally free.
PSTN Fall back
In case of interruption or your Internet service goes down PSTN-Fall back features
immediately switches your system into PSTN mode. It can run on PSTN mode only if you
desire.
Telecommuters
With the help of soft phone an employee can still be in touch with his extension
number and office while traveling outside the office or between the offices. You can
answer your extension from outside you office thru soft phone or even cell phone.
Branch Office & Telecommuter Support
By just getting low cost server for each of you office and configuration thru
control panel you can link your server and can make free office-to-office calling and call
forwarding to any extension in your linked server network via VoIP.
Web Control Panel
Web Control Panel feature comes in two interface administrator to manage
remotely and individual user to manage their personal configurations from remote.
Plug and Play
With pre-configured according to your order it is just a matter of plug and play
and your system is up once you plug in your office for the fist time. Before shipping we
configure and test at our in-house so that it’s just a matter of plug and play at your end.
Powerful Reporting
With the extensive feature of powerful reporting, one can view call logs for all
extensions in real time with search and filter parameters from web admin panel. If you
are using our VoIP network we can provide real time billing so that you can view how
much you’re spending on telephone. With Web User Panel one can look at his call and
also can return calls by just clicking on it.
Call Intercept
Enables group administrators to intercept calls routed to a non-working internal
line with informative announcements and alternate routing options. The service may be
assigned to an individual user’s phone number (e.g., when he is not in the office) or it
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can be assigned to all the members of the group.
Call Pickup
This feature enables a user to answer any ringing line within their pick-up group.
A pick up group is a group administrator-defined set of users within the group, to which
the call pickup feature applies. To pick up a ringing call, a user dials the call pick up
feature code. The user is then connected to the caller. If more than one line in the pick
up group is ringing, the call that has been ringing the first is answered. Users can also
execute call pickup via a web interface.
Hunt Groups
Hunt Groups allow users within a group to be included in a specified sub-group to
handle incoming calls received by an assigned Hunt Groups phone number.
Phone Status Monitoring
Ability to monitor the phone status of users within a group (busy, idle, do not
disturb).
Call Capacity Management
The Call Capacity Management feature enables service providers to limit the call
traffic associated with individual groups by limiting the number of simultaneous calls
that can be made to or from a customer premise.
Call Trace
This feature enables a user to request that a call they have received to be
automatically traced by dialing a feature access code after the call.
Call Waiting
Answer a call while already on another call.
Speed Dial
This feature enables users to dial two-digit codes to call up to 100 frequently
called numbers. Entry of the two-digit code is preceded by a configurable prefix: * , or
#. Users can program the numbers in their directory thru web portal, or directly through
their phone using the respective feature access code (*75 default).
In-bound Call Center
The Call Center is also an Asterisk solution based PBX system. The Call Center Full
Featured A.C.D. (Automatic Call Distribution), which allows you to route incoming calls
to your Users/Agents in many ways like first to pick gets the call, ring who has fewest
calls or least recently called, ring in order/memory, ring one person at a time or ring a
random person, etc. The GUI has a separate window for monitoring and managing the
ACD queues. This shows the number of agents on duty and available, the number of
unanswered calls in the queue, and the average and maximum wait times. The current
status of these statistics is shown, along with a rolling average over several time
periods. Enables business groups to set up a basic call center with incoming calls
received by a single phone number distributed among a group of users or agents.
Unlimited Call Queues (for call centers only)
You can extensively manage your call queues, priorities call of user/agent,
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personalized audio announcement; hold time limits so that the caller does not remain in
your queue for long time, assign length of time before the same person is called again,
setting caller limit to keep the queue volume manageable. Each queue comes with a
variety of options:
Skills-based Routing (for call centers only)
You can assign queue of call according to your priority i.e. pass on more call to
your star performer and less to under performer. It also has additional layer of
prioritization based on agent order so that you can have similarly skilled agents who
receive calls in an ordered round-robin fashion.
Graphical Queue Reports (for call centers only)
You can view complete graphic based reporting like abandoned calls, completed
calls, hold time, average call length, agent productivity, etc. and at the same time view
the graphs of call completion on hourly and weekly basis.
Agent Call Recording /Monitoring / Barge-In (for call centers only)
With this feature you can choose an agent, and select a number of calls to record
and will then record this specified number of calls and automatically stop recording
when the limit is reached. Call center supervisors and managers may monitor calls, and
may optionally record calls, or barge in to calls "on-the-fly" as needed. When monitoring
calls, other supervisors and managers can see at a glance on the graphical user interface
which calls are already being monitored. With call recording, management may record
all calls on a per user basis, or they may start and stop recording during specific calls.
With call "barge-in," supervisors may easily join a call in progress to lend advice or
clarification.
Multiple Auto-Attendants (for call centers only)
Multiple Auto-Attendants feature directs calls to a different numbers for different
departments.
Upload Voice Prompts (for call centers only)
This feature gives you the ability to upload highly professional pre-recorded voice
prompts when simple recordings from your phone (Standard Edition) just won't do.
Alarm Limits (for call centers only)
The call center manager or supervisor can set alarm limits so they will be
instantly notified when the number of calls in the ACD queue exceeds a specified limit,
or when the average wait time is greater than the defined limit.
Computer Telephony Integration (CTI) (for call centers only)
Computer Telephony Integration is the ability of the phone system to interact with
the company or employee computer system(s). An IP-based phone system is particularly
well suited for CTI. Asterisk uses CTI to communicate with Windows, Linux and UNIX
applications.
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Application Integration
Asterisk offers application thru AGI interface.
Similar concept to CGI
Offers choices of programming platforms
Java
PHP
Pearl
Python
CRM integration for all of above
•
CRM Integration
With a few simple clicks, you can make your desk phone (hard or soft) integrate
with your web browser, to make your PC open browser windows for sales automation
system or for ticketing in few seconds.
Outlook Integration and E-Mail Integration
It is fully integrated with Microsoft Outlook! If you want to call someone from
your contact list or from any email in you inbox, just right click and you phone on desk
will ring and you can pick it up and call. When you get incoming call on your phone, a
screen pop up will show the name of the person calling from your outlook contacts.
When receiving a call, the user’s Microsoft Outlook contact database is searched for a
match of the caller’s phone number and prompts on your screen the name and number
of the caller. If you want to add new Outlook journal entries of incoming or outgoing
number, users may choose to have automatically opened for incoming and/or outgoing
calls.
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