SIP Trunking for Asterisk
Choosing Sangoma as Your Asterisk SIP Trunking Provider
We’re so grateful to the open source community for their continued contributions to the Asterisk open source toolkit. We’d like to encourage you to further support our efforts by choosing SIPStation for your SIP trunking needs.
Save money when you choose Sangoma as your SIP provider for Asterisk.
Frequently Asked Questions
Asterisk is a free open source communications framework sponsored by Sangoma. Asterisk can be used to build communications applications to extend an existing PBX, or as an individual solution.
Session Initiation Protocol, or SIP, is an application layer protocol used to achieve a VoIP call between two endpoints. As virtual versions of analog lines, SIP trunks allow for multiple channels to be connected to a single PBX.
As sponsor of the Asterisk open source toolkit, Sangoma also offers a robust portfolio of compatible Unified Communications products and services, including turn-key business phone systems, VoIP gateways, telephony cards, and more.
The most recent release of Asterisk can be downloaded here.