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A leading VSAT Provider in Africa

ASTERISK BASED VOIP SOLUTION                                                                                                                                                                                        
Gotalk! - voice over ipOur 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%.

  1. Supports SIP, IAX and H.323 protocols and can therefore be integrated with many VOIP providers.
  2. Up to 300 simultaneous conference calls depending on the processor power of host machine
  3. Customizable IVR (Integrated Voice Recordings) menus.
  4. Unlimited voicemail support.
  5. Can be easily integrated to traditional PBXs.


  1. Free inter-office calls
  2. Connect to other services including the public switched telephone network (PSTN)
  3. Subsidized rates on international calls
  4. Ability to quickly & efficiently identify network problems.




Other technical specifications are listed in the following table.

Call features

Automated Attendant

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 Transfer

Call Waiting

Caller ID

Caller ID Blocking

Caller ID on Call Waiting

Conference Bridging

Direct Inward System Access

Interactive Voice Response (IVR)

Music On Hold

Music On Transfer:

- Flexible Mp3-based System

- Random or Linear Play

- Volume Control


TDMoE (Time Division Multiplex over Ethernet)

Allows direct connection of Asterisk PBX

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






Remote Call Pickup

Remote Office Support

Roaming Extensions

Route by Caller ID


VoIP Gateways


- Visual Indicator for Message Waiting

- Stutter Dialtone for Message Waiting

- Voicemail to email

- Voicemail Groups

- Web Voicemail Interface

Computer-Telephony Integration

AGI (Asterisk Gateway Interface)

Graphical Call Manager

Outbound Call Spooling

TCP/IP Management Interface [GUI]

Codecs support


G.711 (A-Law & ?-Law)


G.723.1 (pass through)


G.729 (through purchase of a commercial license)







IAX™ (Inter-Asterisk Exchange)

SIP (Session Initiation Protocol)