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.
IP-enabled communications promote collaboration among employees, streamlining work processes and increasing productivity.
Ease of Use
With a user-friendly GUI design and plug-and-play implementation, the SwitchTrunk™ service further reinforces EtherSpeak’s reputation for having the industry’s easiest solutions to deploy and use.
SwitchTrunk can ensure nationwide Quality-of-Service (QoS). With low-latency, high-data throughput, minimal/zero packet loss, and negligible jitter, you can experience the true potential of VoIP.
A managed SIP trunking solution reduces the time spent on network deployment and maintenance, increasing flexibility and reducing OpEx and total cost of ownership.
SwitchTrunk™ easily scales to increase bandwidth as a business grows, addressing small-to-large enterprises with solutions that fit everything from small business to corporate headquarters.
Lessened Administrative Burden
SIP Trunking uses the same Internet connection used for normal data, eliminating the need for PRI/BRI or other connections and equipment.