SIP Trunking for Asterisk

Enjoy native SIP connections to your Asterisk-based platform with SwitchTrunks by EtherSpeak

SwitchTrunk™ is a risk-free approach to SIP Trunking, enabling customers of IP PBX phone systems based on Asterisk* (Switchvox, Fonality Trixbox, AastraLink, etc.) to save on capital expenditures, hardware costs and additional fees typically associated with PRI/BRI, analog voice circuits or even on-premise SIP trunking solutions.

EtherSpeak’s native SIP trunking solution for the Asterisk community is especially engineered for Asterisk-based IP PBX customers. Customers of all sizes can finally make full use of their investment by connecting to fixed-line PSTN via SIP networks. This means customers are able to realize all of the benefits of IP communications without additional hardware or firewall/gateway requirements.

Want to know more about SIP Trunking and IP communication features?

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