[Asterisk-Users] IPManager PBX Features

[Thorben Jensen]

IPManager version 1.6 has just been released. Below is a list of some of the
features you will get on your Asterisk server using IPManager to generate
your configuration files.

 

Download: http://ipsoftware.thorben.dk <http://ipsoftware.thorben.dk/> 

 


PBX Features


The following features will be available to users of the PBX if you are
using IPManager to configure your PBX.

 

*	Unconditional forwarding of all calls to your mobile or other phone
number.
*	Dual calling - your desk phone and your mobile will ring
simultaneously and whatever phone is answered first will get the call.
*	Do Not Disturb until a predefined time - You can set DND from your
extension to a predefined time - if set to 10:15 all calls to your extension
will be told that you are not available until 10:15. The DND will
automatically be removed at 10:15.
*	All Extensions can have opening/closing hours this includes Phones,
Queues, IVR-menus and Direct Dial In's. This means that you can have certain
hours that an extension is open. Any phone you can override that setting by
dialling a code. It is possible to configure an alternative extension to go
to when an extension is closed. Examples:

*	You have a Queue that callers are sent to during opening hours and
when the queue is closed all calls are sent to an IVR-menu.
*	Your own extension is open from 9:00-12:00 and 13:00-17:00 all other
times calls are sent to your mobile.
*	You have a main IVR-menu that asks caller to press 1 for sales, 2
for service during opening hours and another IVR-Menu that tells callers
that you are closed but they can press 1 to have you call them back (You
will get an e-mail), or 2 to leave a message on the answering service..
*	All calls to your main queue are redirected to your mobile when
closed.

*	Virtual Users - You can have a Virtual number and that means that
you can login at any extension and you will receive all your calls at that
extension. Callers will also be sent to your Voicemail (if configured) and
you will be using your own caller ID when making calls from that extension.
This can typically be used by people travelling between departments or shift
workers sharing the same desk. When you are not logged in at any extension,
you can have your call sent to any other phone number - ex. your mobile.
*	Wake-up calls - Schedule calls to your extension at scheduled time.
*	Voicemail - you will have any messages left in your voicemail
forwarded to your e-mail account with a .wav file that holds your message.
Just click the .wav file and you will hear the message.
*	The LED's will work on SNOM phones (maybe also other phones but
that's not tested).
*	Limiting the number of active incoming/outgoing lines on your server
- it's better to give a busy signal when you run out of bandwidth than
everybody gets poor speech quality because of bandwidth limitations. 
*	Speed Dialling - You can define speed dial number that everybody can
use for ease of dialling. Ex. dial "1" to and the server will dial
"00353469071189". You can have unlimited Speed Dials - Use IPSpeedDial
application for easy maintenance of speed dial numbers.
*	Queue Identification - see on your phone which queue the caller
entered as well as caller ID.
*	Music Groups - You can define different music groups for each queue.
It's easy to upload MP3-files to your server.
*	Any extension can be added to a queue - you can even have mobile
phones receive calls from a queue. It is possible to give priority to the
phone members of a queue meaning that lower priorities will receive calls
only when higher priorities are busy.
*	Possible for callers to break out of a queue and ask to be called
back - you will receive an e-mail with the number to call back.
*	Any extension can be dialled directly by using DID's.
*	Virtual faxes - will receive a fax and send it to your e-mail
account.
*	DISA - Direct Inward System Access - Dial into your server and get a
second dial tone and make a call out to the cheap VoIP rates - Password
protected and you can have different password for each user.
*	Call Back to predefined extension - Have your customers call into
your server and the server will call back to the caller and connect him to a
predefined extension (this would typically be a queue). Great savings if you
are calling from a mobile abroad this will typically save you between 40-75
% in roaming charges. It is typically a LOT cheaper to call a mobile abroad
than having the mobile make the call.
*	Call Back and connect to any number - Call your server and dial a
password. You can now dial the number you want to be connected to and the
server will hang up and call you back and connect you to the number you
requested. This way you are calling at the cheap VoIP rates.
*	Conferencing - Dial your server type a conference number and have a
conference with as many people as your broadband line can handle.
*	Call Me Back Service - Send a caller to this extension from a Queue
or IVR-Menu and the caller will be asked to dial the number they want to be
called back at. You will receive an e-mail with the number requested.
*	Queues can have these features:

*	Tell the caller their position in the queue at predefined intervals
*	Tell the caller the approximate waiting time in the queue
*	Tell the operator how long the caller waited in the queue
*	Have periodic announcements. This is typically used for advertising
or giving caller the opportunity to break out of the queue by dialling a
number.
*	Have different background music depending on the queue they are
waiting in.
*	You can choose many different ring strategies:

*	Ringall - calls all members at the same time
*	Round Robin - Starts at one extension and continues with all others
in a circular fasion.
*	Round Robin memory - Starts at one extension and continues with all
others in a circular fasion and remembers where it got to and starts from
that member the next call.
*	Least Recent - The member that has been available the longest will
get the call.
*	Fewest calls - The member that has received the fewest calls will
the the call.
*	Random - the server randomly selects which member should get the
call.

*	Timeout - how long you will allow the callers to be in the queue
before sending the caller to an alternative extension.

*	Least Cost Routing - Direct your calls to the cheapest provider for
the destination. You can even specify different routes for each extension or
group of extensions; this can be used for separate billing per department
etc.
*	Transfer directly to any voicemail - use if the caller wants to
leave a message on somebody's voicemail.
*	Speaking Clock - tells date and time of your server.
*	Echo Test - Test what you phone sounds like.
*	.and much more. New features are being added continuously and if you
need a features not listed here just send me an e-mail (ipm at thorben.dk) and
I will see if I can implement it. 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051016/eeeffda7/attachment.htm