December 7, 2010

Open source VOIP Phone Systems

iCepts offers an open source business phone system which utilizes VoIP (Voice over IP) technology.

No Charge!

  • There is no charge for the base software.
  • There is no charge for add-on features.
  • There are no annual licensing fees.
  • There is no charge per user.
 You only pay for:

  • Hardware
  • Services to install
  • Your phone lines
  • That’s it!

Asterisk Open-Source VoIP Phone System Features   

Download VOIP Case Study

Case Study Front PageHTML tutorial

 

“Open Source” Voice Over IP (VoIP) explained:

The hardware required is industry standard, not proprietary.  So the cost of implementing this open source voip phone system is just hardware (server, phones, switches) and labor costs – no expensive license fees.  For fractions of the cost of a big-name VoIP System, iCepts is offering a fantastic phone system with some of the same (or better) features.  And because these IP Telephony engines are released as Open Source software (providing access to the underlying source code), there is worldwide development and support, while being sponsored by for-profit corporations.  These can be utilized with any number of phone extensions – from only a few up to several hundred.

The Open source VoIP systems that we are proposing still utilize the traditional phone lines from carriers (Verizon, Sprint, AT&T, etc.) that a company already has in place.  When using these telephone circuits, local or long-distance calls aren’t being sent “over the Internet” (except in cases of branch access to the phone system via VPN).  However, there is also the capability to utilize Internet Telephone Service Providers either as a backup to traditional Telco lines, or to replace them, often at a very substantial cost savings*.  Again, these Internet telephone providers are only an option, as traditional lines, such as T1’s or analog (POTS) lines can continue to be used.

We also have the capability to interface this phone system (or another Asterisk based phone system) to the software packages that we support, including Microsoft Dynamics NAV (Navision) and others.  With this Click-to-Call interface, we can allow employees to place calls to customers or vendors (or other contacts w/ phone numbers) directly from the ERP application itself.  For example, a customer service representative could be in an order screen in Microsoft NAV and just click the Click-to-Call button.  His/her desktop phone (or softphone) rings and the phone system automatically dials and connects the customer/vendor call to the phone of the representative.

At iCepts, we are an authorized HP dealer for server-grade platforms, an authorized Sangoma partner for telephony cards (the interface to Telco lines), and an authorized Polycom reseller for their desktop phones.

We can put in a pilot/prototype on your hardware using an Internet telephone company that would allow you to try the system without interrupting your existing phone service.  This gives you a low-cost opportunity to check out features without taking down your existing phone system.

*Note: Acceptable voice quality depends on having proper & adequate bandwidth to the Internet.

Asterisk’s Open Source Voice Mail Greetings User Portal:

This is an add-on to the Open Source Asterisk VoIP User Portal, written by iCepts Technology and released to the Open Source Community, which allows the creation and management of Named greetings for voicemail.  Named greeetings are saved greetings recorded by the end-user, who then can use these greetings as one of their system greetings (“The Busy”, “Unavailable”, or “Temporary” greetings) for their voicemail mailbox.  This allows users to change their system greetings from a browser, without having to re-record these greetings:  Click Here for more information, screenshots and instructions to set up your named greetings voicemail.

Asterisk Open source VOIP Features

In addition to being a “feature rich” phone system, Open Source VoIP can save your small company money by not having any charge for the base software, no charge for add-on features, no annual licensing fees, plus there is no charge per user!

 

 

Call Features

ADSI On-Screen Menu System
Alarm Receiver
Append Message
Authentication
Automated Attendant
Blacklists
Blind Transfer
Call Detail Records
Call Forward on Busy
Call Forward on No Answer
Call Forward Variable
Call Monitoring
Call Parking
Call Queuing
Call Recording
Call Retrieval
Call Routing (DID & ANI)
Call Snooping
Call Transfer
Call Waiting
Caller ID
Caller ID Blocking
Caller ID on Call Waiting
Calling Cards
Conference Bridging
Database Store / Retrieve
Database Integration
Dial by Name
Direct Inward System Access
Distinctive Ring
Distributed Universal Number Discovery (DUNDi™)
Do Not Disturb
E911
ENUM
Fax Transmit and Receive
Flexible Extension Logic
Interactive Directory Listing
Interactive Voice Response (IVR)
Local and Remote Call Agents
Macros
Music On Hold
Music On Transfer:
– Flexible Mp3-based System
– Random or Linear Play
– Volume Control

Predictive Dialer
Privacy
Open Settlement Protocol (OSP)
Overhead Paging
Protocol Conversion
Remote Call Pickup
Remote Office Support
Roaming Extensions
Route by Caller ID
SMS Messaging
Spell / Say
Streaming Media Access
Supervised Transfer

Call Features

Talk Detection
Text-to-Speech (via Festival)
Three-way Calling
Time and Date
Transcoding
Trunking
VoIP Gateways
Voicemail:
– Visual Indicator for Message Waiting
– Stutter Dialtone for Message Waiting
– Voicemail to email
– Voicemail Groups
– Web Voicemail Interface
Zapateller

Computer-Telephony Integration

AGI (Asterisk Gateway Interface)
Graphical Call Manager
Outbound Call Spooling
Predictive Dialer
TCP/IP Management Interface

Scalability

TDMoE (Time Division Multiplex over Ethernet)
Allows direct connection of Asterisk PBX
Zero latency
Uses commodity Ethernet hardware
Voice-over IP
Allows for integration of physically separate installations
Uses commonly deployed data connections
Allows a unified dialplan across multiple offices

Codecs

ADPCM
G.711 (A-Law & μ-Law)
G.719 (pass through)
G.722
G.722.1 licensed from Polycom®
G.722.1 Annex C licensed from Polycom®
G.723.1 (pass through)
G.726
G.729a
GSM
iLBC
Linear
LPC-10
Speex

VoIP Protocols

Google Talk
H.323
IAX™ (Inter-Asterisk eXchange)
Jingle/XMPP
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)
SIP (Session Initiation Protocol)
Skype
UNIStim

Traditional Telephony Protocols

E&M
E&M Wink
Feature Group D
FXS
FXO
GR-303
Loopstart
Groundstart
Kewlstart
MF and DTMF support
Robbed-bit Signaling (RBS) Types
MFC-R2 (Not supported. However, a patch is available)

ISDN Protocols

AT&T 4ESS
EuroISDN PRI and BRI
Lucent 5ESS
National ISDN 1
National ISDN 2
NFAS
Nortel DMS100
Q.SIG