SIP Trunking for Asterisk
SIPStation for Asterisk
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.
Start Saving in Minutes
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.
Bring New Ease to Connectivity
SIP Trunking Features
- Universal – Works with any SIP or SIP enabled PBX.
- Remote Call Forwarding (RCF) – If your SIP trunk cannot deliver a call to your PBX, it can be routed to another destination (such as an analog line, or cell phone).
- On Demand Capacity – With Concurrency Bursting, you won’t risk rejecting calls due to limited capacity, or pay for connectivity you won’t use.
- Phone Numbers – Get local or Toll Free phone numbers across the US and Canada.
- Number Porting – Bring your numbers with you.
- Migration – Easily connect to a legacy system with a Digium or Sangoma gateway.
- Reliability – Redundant SIP trunks help ensure your services are always up and working.
- Trunk Groups – Share your SIP trunk across multiple locations to save even more!
Channelized Rate Plan
Perfect for businesses that prefer a set, predictable monthly phone bill.
- Unlimited inbound and outbound local and long distance calls on a per channel per call basis.
- Channels can always be added for more capacity.
* Inbound and outbound calls available in contiguous US 48 states