IP Telephony

IP Telephony

Many organisations migrate to IP-based networks to take advantage of the cost savings gained from converging voice and data traffic onto a single network structure. Initas builds converged networks that cater for the service availability and traffic prioritisation requirements when running voice, video and data traffic across one network.

Initas migrates, integrates, secures and manages IP Telephony, helping customers to mitigate the risk of end-of-life PBXs and reducing the operational costs associated with managing and administrating multiple networks. We enhance the value of telephony with telephony management systems such as billing and call management and security assessments, to reduce the risks of telephony misuse and fraud.

Based on the award winning Asterisks®, the architecture of our IP Telephony solutions are open and extensible, and allows customers to incorporate custom-developed classes, which enables developers to extend the IP solution to meet unique customer business needs.

About Asterisk

Asterisk® is the leading open source telephony project. Originally written by Mark Spencer of Digium, Inc., Asterisks is currently boasting over two million users and supports a wide range of telephony protocols. Suse Linux It includes rich support for the handling and transmission of voice over traditional telephony interfaces including analog lines, ISDN-BRI lines and digital T1/E1 trunks. Asterisk also features support for a wide range of VoIP protocols including SIP, IAX and H.323 among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

Among the many applications you can create with AsteriskNOW are:

  • VoIP Gateway
  • Skype Gateway
  • IP PBX
  • Call Center ACD
  • Conference Bridge
  • IVR Server
  • Voicemail System
  • Call Recorder
  • Fax Server
  • Speech Server

Contact us to discuss how our IP Telephony solution will help your organistion.

Asterisk was originally built as a PBX and today represents an astonishing 18% of global market for business telephone systems. The base feature set includes many of the most popular and powerful PBX functions.
Tapping the power of Asterisk requires some knowledge of Linux, telephony, basic script programming and IP networking. We also offer Digium’s Switchvox, a complete IP PBX system based on Asterisk.

Our IP IVR removes the constraints enforced by legacy PSTN circuit-based IVR applications by IP-enabling IVR applications and creating connections to Web-based content. Based on the award winning Asterisks®, the architecture is open and extensible, and it allows customers to extend the IP IVR solution to meet unique customer business needs.

Call Centre ACD
Automatic Call Distributors (ACDs) allow call centers to handle thousands of simultaneous calls, routing them to agents based on caller input, dialed number, load and other factors. ACD systems typically cost tens if not hundreds of thousands of dollars and require specialized training to install and operate. With Asterisk you can build a powerful ACD for the cost of the server hardware and phones.

Gateways connect legacy phone equipment (PBXs, ACDs, voicemail systems, etc.) to modern VoIP systems and services. A VoIP Gateway enables direct routing between IP, digital, analog and GSM networks. With these devices companies can significantly reduce the money they spend on telephony.
The core idea behind cost saving with VoIP GSM Gateways is Least Cost Routing (LCR).Through least cost routing the gateways select the most cost-effective telephone connection. They check the number which is dialed as well as rate information which is stored in an internal routing table. Because several SIM cards, GSM modules and PSTN lines are integrated within the VoIP Gateway it is able to make relatively cheaper GSM to GSM calls instead of expensive IP to GSM calls.
Asterisk supports many different communications protocols from both the modern world of VoIP and from the legacy PSTN. This makes it a powerful tool for building gateways and protocol converters.

Conference Bridge
Asterisk’s “MeetMe” conference bridge application allows you to build sophisticated multi-party conference applications with only a few lines of Dialplan script. Capable of scaling to hundreds of parties, Asterisk-based conference servers represent one of the most compelling values of Asterisk.

Asterisk’s integrated voicemail application allows integrators to replace proprietary legacy voicemail systems with Asterisk-based solutions at a fraction of the cost. Asterisk’s voicemail can be implemented as a basic stand-alone system or can act as the front end to a Unified Messaging system, storing message using IMAP or ODBC. Building a voicemail server is simple and requires only a bit of dialplan scripting.