Ever wanted to build you own VOIP (voice over internet protocol) network like Skype or Fring? Then this might be a free opportunity for you to start from using an open source and widely used internet telephony software called Asterisk. All funds required on your part will be for the hardware and bandwidth cost. Asterisk is an Opensource internet telephony solution to cater all your PBX , Gateway, Voicemail, IVR, Conference, ACD needs for SMB’s (Small medium businesses). Asterisk open source code is widely used by some of the big names including Google too.
So how to connect to different telephones and exchanges using Asterisk? Answer is simple, using SIP (Session Initiation protocol) Proxy Server.
Some Open Source SIP Proxys
Excellent open source SIP Proxys are available on the Internet. Check
- FreeSwitch: SIP Proxy and PBX
- SIP Express Router: SER is used by many SIP providers standalone or in conjunction with Asterisk
- sipX from Sipfoundry is a native SIP proxy but also a complete SIP PBX
- OpenSER – scalable and robust SIP server with TLS support
Once your SIP proxy server is ready you can use SIP channel Module to make a connectivity between SIP Proxy server and Asterisk Server to use asterisk functionality for a wider range of things.
SIP Channel Module
- a SIP client: This means that Asterisk registers as a client to another SIP server and receives and places calls to this server. Incoming calls are routed to an Asterisk extension.
- a SIP server: Asterisk can be configured so that SIP clients (phones, software clients) register to the Asterisk server and set up SIP sessions with the server, i.e. calls and answers incoming calls. This said, Asterisk is not a full-feature SIP server like SIP express router or OpenSER. If you are going to have thousands of SIP phones, you should use SER or OpenSER and forward calls to Asterisk for voicemail or PSTN access.
- a SIP gateway: Asterisk acts as a Media gateway between SIP, IAX, MGCP, H.323 and PSTN connections. As an example, an Asterisk server can be connected to ISDN to give your SIP clients connectivity to the switched telephony network.
Hope our readers can build their own communication servers one day using open source technologies. There is loads of help available to download, install and configure SIP Proxys and Asterisk for different roles. Anyone can just google out them.