Digium has released the latest version of its Digium SIP trunking product, available via its Digium Cloud Services agents.
Digium SIP trunking offers VoIP connectivity for Switchvox, Digium’s unified communications (UC) system; the Asterisk open source communications software; or any IP-enabled phone system.
“Digium’s award-winning Switchvox Cloud offering is deployed on a high-availability, fault-tolerant network. Now, that same network is available for anyone who needs SIP trunking capability,” said Mark Amick, director of product management at Digium. “Digium has ensured maximum reliability by building a carrier-class network and is excited to offer our Switchvox customers and the Asterisk community the only SIP trunking network built and certified by the Asterisk company. Additionally, Digium SIP Trunking isn’t limited to just Asterisk users, it’s based on open standards and will work with any RFC standard, SIP-enabled PBX.”
Digium SIP Trunking has no setup fees and is available in two plans: the channelized plan offers unlimited usage at a flat rate; the metered plan is pay as you go, and available for customers that need flexibility due to seasonality or unpredictable call volume.
Digium SIP Trunking also provides unlimited free calls between DCS customers.
“Digium’s SIP trunking offering is more than simple dial tone; it’s a fully featured offering including nationwide DIDs, 911 enabled DIDs and toll-free DIDs,” the company said. “Digium can also port both local and toll-free numbers from other carriers, allowing customers to bring their existing numbers with them. With support for multiple codecs and T.38 fax pass-through, customers can rest assured that they will be able to use their bandwidth efficiently while maintaining quality of service for both voice and fax.”
