Features supported by Asterisk – The Open Source VOIP Softswitch


Asterisk, is a freely available open source IP Telephony (VOIP) Soft Switch. In this article, let us look at some of the features supported by Asterisk to evaluate its capabilities, when used along with IP Phones and PSTN/SIP Trunks.

  • Call Handling Functions: Asterisk supports all the normal call handling functions like call pickup, call terminate, call hold, call park, blind transfer, call waiting, Caller ID, Direct Inward Dialing, etc.
  • Call Monitoring/ Snooping: Asterisk allows a super administrator to listen to any calls on any of the extensions. It is also possible for them to barge-in in to the call if there is an important high priority message to be given to a person who is already on another call.
  • Call Detail Recording: Asterisk has in-built capabilities to record all the calls/ duration/ extension numbers etc, and export it  in a file format,  so that the administrators can monitor the call usage per user/ per department statistics in their company.
  • Simple/ Meet Me Conference: Its possible to do a simple conference call by holding a line and calling another number to make a conference; Its also possible to create a Meet Me conference where a conference number could be advertised and people need to dial in (if required with a pass code) in order to enter the conference.
  • Direct Inward System Access (DISA)
  • Music On Hold: When a caller is kept on hold, music is played by the PBX. Its possible to select which music file needs to be played, using Asterisk.
  • Voice Mail: If a person is not available in their extension, its possible to divert the incoming call to that extension to a voice mail after a certain number of unanswered rings, where the caller can leave a recorded voice message. Its also possible to email voice mail notifications, to the extension user, as soon as they are left. Its possible to create a Do Not Disturb policy so that the calls are transferred directly to the voice mail instead of ringing on the extension, using Asterisk.
  • Dual Ringing: A call, with appropriate set-up, can be made to ring both in the desk phone as well as cell phone (for the same employee). It gets connected to which ever device picks it up first.
  • IVR/ Automated Attendant: Its possible not only to give a customized welcome prompt to callers, but its also possible to create complex IVR rules to make Asterisk fetch information from a Database and give it to the users requesting it, preferably after authenticating them. Database Integration with Asterisk is possible.
  • ITSP: These days, its cheaper to make long distance calls using VOIP service providers over the Internet. These service providers are called ITSP – Internet Telephony Service Providers and their SIP trunks can be integrated with Asterisk. Its possible to use Least Cost Routing to determine if its cheaper to route a call through the Analog/Digital Trunk lines (or) SIP Trunks automatically using Asterisk and route it over the cheapest medium.
  • SIP Trunking between branches: If Asterisk is deployed in multiple branches of the same organization, its possible to route all the inter-organization calls over the Internet using SIP Trunking facility in Asterisk in order to save on phone bills to other branches.
  • Call Recording: Its possible for Asterisk to automatically record calls made from certain extensions, for example.
  • Email Client Integration: Its possible to integrate Asterisk with corporate directories like LDAP and Email Clients like MS Outlook which allows them to just click on a contact to dial their extension from PC.
  • Soft Phone Support: Asterisk supports freely down-loadable SIP based Soft Phones, which can be used with a PC/ headset to make calls.
  • No limit to number of extensions: While most of the commercial IP Telephony vendors have licensing costs for every additional extension to be added in an IP PBX, there is no such licensing with Asterisk. You can add any number of extensions (as long as the hardware configuration of the PC/Server supports it).
  • Termination of Analog/Digital Trunk Lines and Extensions: Its possible to connect Analog Trunks (FXO Lines) and Digital Trunks (ISDN, PRI lines) and also analog phones with Asterisk using Digium/ Sangoma cards which connect to the PCI interface of the PC/Server where Asterisk is running.

This is just a glimpse of the important features supported by Asterisk. There are many more telephony features supported by Asterisk than any commercial / Analog-Digital PBX. Do consider implementing Asterisk VOIP system in your organization if a good implementation partner is available in your region. Asterisk is a CLI based implementation, but there are GUI based alternatives available here and here.

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