Digium, the sponsor and maintainer of the Asterisk project, offers high quality, cost-effective SIP trunking for your Asterisk server, Switchvox, or virtually any IP PBX. Digium SIP Trunking is now powered by SIPStation, a low-cost, feature-rich telephony service available across the US and Canada.
Deploying SIPStation SIP trunking in lieu of traditional phone lines greatly reduces an organization’s telephony costs while implementing a higher standard of service through your Asterisk server or any other IP PBX. Replacing terminated T1-based PSTN connections with SIPStation SIP trunks to an ITSP simplifies communication resources into a single data connection and enhances voice quality. SIP trunks lower your company’s rates instantly for local and long distance calls and allows companies to save thousands off their phone bills.
Perfect for businesses that prefer a set, predictable monthly phone bill.
* Inbound and outbound calls available in contiguous US 48 states