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Say hello to Asterisk


The Open Source Communications Platform

The days of expensive proprietary telecom software are over. Now there's the open technology of Asterisk.

Since Asterisk is Linux-based, it inherits all of the power and stability of the operating system. Linux provides open source alternatives to proprietary applications. Asterisk is the first package to fit all telecommunication needs in a broad variety of environments with advanced features.

The name Asterisk is derived from the all-inclusive "wildcard" symbol in UNIX, because the Asterisk platform is providing opportunities for developers worldwide to create solutions which would otherwise be cost-prohibitive or impossible.

Asterisk is global, supporting all sorts of applications, like call centers, IVR platforms and VoIP switches. There are numerous advantages to such flexible software as it can be easily customised to meet any need.

Advantages of using Asterisk

There are many companies offering phone systems or PBX solutions in South Africa. Added to this, these systems can often be very technical and therefore it can be hard when deciding what solution is best suited to your company's needs. Here are a few reasons why you would want to choose an ASTROVOICE Asterisk PBX.

Where can it be deployed?

Asterisk is extremely flexible software and since its code is Open Source and freely available it can be easily customised to meet any need.

Here are some examples of organisations where asterisk be deployed:

  • Contact/Call Centres
  • Hotels
  • Small - Medium sized business
  • Large Organisations
  • Companies with multiple offices
  • Universities/Colleges

Cost Savings

One of the main benefits of using an Asterisk phone system is cost savings. Asterisk is an Open Source project which means there is no software or licensing costs. It works with low-cost Linux telephony hardware but can also be integrated with most of the existing hardware used by South African companies today. This means that switching your phone system to Asterisk does not require a large initial cash outlay but rather can be integrated with your existing phone system. The phone system will work with traditional Telkom phone lines (PSTN/ISDN) aswell as with VoIP, calls can be routed through which ever way is cheapest for your organisation. ASTROVOICE have a a full range of Least Cost Routing solutions available. Asterisk is also very scalable and can be easily expanded, therefore the cost of adding a new user is small compared to that of a proprietary system.

Customisations

Asterisk is an extremely flexible Open Source application. It works with all major industry standards so calls can be made through standard phone lines or routed over the Internet. Also since Asterisk is an Open Source project and its code is freely available we can easily customise it to meet your needs. Advanced solutions such as Auto Diallers and complex IVR's can be developed through Asterisk.

Scalability

Many proprietary PBX systems limit the amount of users that can be connected to the system or else the cost of adding a new user can often be very high. This is not the case with Asterisk. Asterisk is extremely scalable, a relatively small server could handle anything between 1 and over a 100 users. The cost of adding a new user is small and can be done quickly. Also if your organisation has more than one location your phones can be linked to one central sever using a VPN or each location can have its own asterisk server which is then linked together.

Support

ASTROVOICE have experienced and highly qualified technications. We are a South African company so support is local and can be delivered quickly. Also Asterisk can be configured and maintained remotely so any customisations or problems that may arise can be done by us off site.

Asterisk Applications

PBX

Asterisk allows you to create a IP and TDM PBX that rivals the features and functionality of traditional telephony switches. Other PBXs are expensive and proprietary. Asterisk is cost-effective, low-maintenance, and flexible enough to handle all voice and data networking. With Asterisk, Digium hardware, and a common PC, anyone can replace an existing switch or complement a PBX by adding VoIP, voicemail, conferencing and many other capabilities. Asterisk integrates with most standards-based IP telephone handsets and software. Analog phones and ADSI screen phones are also supported, proven through installations around the globe, Asterisk is solid and stable in SOHO or Enterprise environments.

Interactive Voice Response (IVR)

Asterisk's flexible IVR capability allows a user to interact with a database using a menu of pre-recorded voice-clips. Using MySQL and other popular databases, Asterisk can interact with the caller through touch tone inputs, record responses, query databases, and utilize AGI scripts to perform specific tasks.

For example, a customer can authenticate a pre-paid calling card with a PIN queried from a database. The Asterisk IVR will give the number of remaining minutes and later disconnect if that customer runs out of time. The spooling feature can allow Asterisk to dial a list of numbers from a database to give warnings during homeland security emergencies. Asterisk IVR allows developers to create a myriad of IVR solutions.

Auto-Attendant

Asterisk's auto-attendant features include greetings, extended greetings, music-on-hold, voice message forwarding and message appending. Asterisk plays music or pre-recorded messages to customers on hold. Music can be sorted into various folders. Separate auto-attendant feature sets can be used for different situations. The voicemail tree supports directories by department, employee, extension, etc., offering flexibility and allowing a small company to appear large. Unbound by the limits of traditional voicemail, Asterisk can support an unlimited number of simultaneous ports.

Conference Bridge

The Meet Me Bridge is fully integrated into Asterisk and supports features essential for business conferences, saving the Asterisk user from what was once a huge expense. The Conference Chairperson can select a listen only or a talk and listen conference. When the Chairperson hangs up the other parties are disconnected. Conferences may be securely accessed only through a pre-defined PIN.

Many companies have also created chat services based on Asterisk Meet Me, with many chat rooms that users can transfer between.

Media Server

Asterisk augments existing PBXs and Gateways with select features for either PSTN or IP protocols. Acting as an adjunct to a legacy system or soft switch, Asterisk can extend features and functionality by providing voicemail and conferencing services. Asterisk can also retrofit traditional TDM PBXs with VoIP extensions to remote offices which appear as normal extensions of the PBX.

VoIP and Protocol Gateway

Asterisk's broad support of both traditional TDM and VoIP protocols permits the construction of flexible gateways between different channel types. Using Asterisk, it is not only easy to create many common varieties of protocols converters, translating between T1, E1, PRI, SIP, IAX, GR-303, MGCP, FXS, and many others. It enables you to create more sophisticated gateways and gateways with redundant links. For example, an MGCP to SIP gateways with a PRI backup can be created in case SIP trunks are unavailable – the possibilities are nearly endless.

VoIP Switch

Asterisk can act as a soft switch in addition to acting as a traditional TDM switch, allowing it to control a variety of devices including phones, gateways, media servers, and other asterisk servers. It can handle virtually any VOIP protocol, including SIP, IAX, H.323, MGCP, and Skinny.

Asterisk collects call detail records and provides a variety of billing options (including Open Settlement Protocol) and may be configured to carry media (especially useful for SIP+NAT situations) or to have devices send media directly to one another. Asterisk adds extra IP or PSTN capabilities to existing PBXs and Gateways. Asterisk extends features and functionality by providing voicemail and conferencing services when acting as an adjunct to a legacy system or soft switch. Asterisk can also add remote VOIP office extensions to traditional TDM PBXs which appear as normal extensions from the pre-existing PBX.

Asterisk Features

Like any PBX, Asterisk allows a number of attached telephones to make calls to one another, and to connect to other telephone services including the PSTN (normal Telkom phones). However Asterisk contains many features which up until now have only being available in expensive proprietary phone systems.

What equipment can be connected?

Asterisk can support standard analog phones and phone lines, ISDN lines and phones and also ISDN30 lines. On top of that it supports IAX2 and SIP, which are VoIP protocols, that can be used to connect phones and also to VoIP providers.

Voice mail

Voice mail is stored as audio files on the Asterisk system. The voice mail messages can be retrieved by dialling the voice mail number, via a web interface or can be send to the user by email. Email notification without attached voice mail is also possible.


Separate messages for busy and unavailable can be recorded.

Music on hold

Music on hold is provided by storing MP3 files to the Asterisk system.

Call parking

Calls can be parked either by putting them on hold or by transferring them to a parking lot system, where they can be retrieved from any phone by dialling the parking slot number, which was announced, when you parked the call. If the call not is retrieved within a predefined time out, it will be send back to the extension that parked it.

Call waiting

Call waiting provides an indication, that you have another call waiting, while you are on the phone already. You can then park your current call to pick up the new one.

Blind transfer

A blind transfer is, where you transfer a call and hang up before the person you transferred the call to has taken the call.

3way transfer

In a 3 way transfer, you put the call on hold and call a second person, talk to the second person and then transfer the call to that person or retrieve the call again.

Call forwarding

You can forward your calls to another extension based on certain conditions: always, on busy, if the call rings out or if you are marked as DND (Do not disturb).

LCR (least cost routing)

Asterisk provides the ability of sending calls to different lines based on the phone no., this can be handy in cases you are using more than one provider and want to use the cheapest route.

Caller-ID

The phones will present the phone no., of the person calling you and a name can also be assigned to phone numbers. This can be done individually in each phone via the local phone book or on request system wide.

DID (Direct Inward Dialling, also called DDI sometimes)

Each user can have a phone number assigned that allows external callers to call him directly instead of going through the receptionist.

Call groups/Call pick up

It is possible to create call groups for people and to pick up calls ringing on other phones within the call group.

Auto Attendant (IVR = Interactive Voice Response)

The IVR function of Asterisk allows options like choosing what department you want to talk to, selecting product groups or a corporate directory, if requested. Other options are possible, too.

Asterisk Feature List

Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records.

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
  • Do Not Disturb
  • E911/E999/E112
  • 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

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 installation
    • Uses commonly deployed data connections
  • Allows a unified dialplan across multiple offices
    • DUNDi Peer-to-peer Call Routing and Discovery

Codecs

  • ADPCM
  • G.711 (A-Law & μ-Law)
  • G.723.1 (pass-through)
  • G.726
  • G.729 (through purchase of 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 and FXO
  • GR-303
  • Loopstart, Groundstart, Kewlstart
  • MF and DTMF support
  • Robbed Bit Signalling (RBS) Types

PRI Protocols

  • 4ESS
  • BRI (ISDN4Linux)
  • DMS100
  • EuroISDN
  • Lucent 5E
  • National ISDN2
  • NFAS
 
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