ASTERISK BASED VOIP SOLUTION
Our Asterisk VoIP (Voice over IP) based solution is a complete Linux based IPBX solution. Because of the process priority that Asterisk must take for quality voice transmission, you will want to run Asterisk from its own box. To this end – we recommend having your own Asterisk IPBX server installed at your head office for a centralized control IPBX
The Asterisk IP VoIP Solution combines complete IP PBX features with standards based SIP for telephony solution that can stand alone or integrate with existing phone systems. Asterisk IP VoIP Solution is an embedded system with built-in SIP Proxy Server and NAT router for SOHO user which not only provides network function but also provide an IP PBX feature in the office.
The setup will utilize the existing bandwidth from your current provider. This would serve all the offices efficiently and will lower all your International telephony costs immediately by more than 50%.
- Supports SIP, IAX and H.323 protocols and can therefore be integrated with many VOIP providers.
- Up to 300 simultaneous conference calls depending on the processor power of host machine
- Customizable IVR (Integrated Voice Recordings) menus.
- Unlimited voicemail support.
- Can be easily integrated to traditional PBXs.
Benefits
- Free inter-office calls
- Connect to other services including the public switched telephone network (PSTN)
- Subsidized rates on international calls
- Ability to quickly & efficiently identify network problems.
Other technical specifications are listed in the following table.
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 (3rd Party OSS Package)
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
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
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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 & ?-Law)
G.722
G.723.1 (pass through)
G.726
G.729 (through purchase of a commercial license)
GSM
iLBC
Linear
LPC-10
Speex
Protocols
IAX™ (Inter-Asterisk Exchange)
H.323
SIP (Session Initiation Protocol)
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)
Traditional Telephony Interoperability
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)
PRI Protocols
4ESS
BRI (ISDN4Linux)
DMS100
EuroISDN
Lucent 5E
National ISDN2
NFAS |
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INVESTMENT SUMMARY
Ref |
Description |
Qty |
Unit Price in USD |
|
Equipment & Installation |
|
|
1. |
SIP Adapter for Voice (1ports ATA) |
3 |
$85.00 |
2. |
Asterisk PBX Server Setup & IP Configuration |
1 |
$1,000.00 |
3. |
Postpaid International VOIP account deposit |
N/A |
$300.00 |
Commercial Terms and Delivery Schedule
Installation:
This has been accounted for in quote.
Validity of Offer / Delivery Timescale:
The terms of this offer remains effective for a period of thirty (30days) from the date of submission, upon which shall have to be reconfirmed in writing.
Please note that the delivery, implementation and commissioning is within seven days (7 days) from the date of contract award.
Service Fee:
Equipment costs and Installation charges to be paid in advance
Warranty:
12 months on all electronic equipment.
Out of Nairobi:
Transport, accommodation and meals to be borne by client
Disclaimer:
Intersat Africa Ltd cannot accept liability for indirect or consequential financial losses.
VAT
The above figures are subject to 16% VAT.
