SAN DIEGO, CA (PRWEB) July 5, 2005 - to medium-
sized businesses (SMBs), today announced enhanced functionality for its flagship IP PBX phone system. Adding support for the Inter-Asterisk exchange (IAX) protocol, the Switchvox system now offers customers more bandwidth savings than traditional protocols, as well as ease of interoperability with additional VoIP providers.
The Switchvox system enables SMBs to easily and affordably create and manage their phone system, using traditional analog lines, as well as VoIP services. Based on customer demand, Switchvox added the IAX protocol, in addition to its current Session Initiation Protocol (SIP), to help businesses reduce set up time, training and support.
ÂIAX was created to a bring level of simplicity and reliability to VoIP systems that other protocols could not,Â said Mark Spencer, CEO of Digium and creator of Asterisk. ÂBy supporting IAX, Switchvox is helping reach out to the large number of Asterisk users who prefer the benefits of IAX.Â
In addition to IAX support, Switchvox now offers other key benefits in its standard phones system, including the following:
- Geographical call routing Â calls can be routed to specific employees based on the prefix of the incoming caller ID, helping to direct calls faster and with fewer steps
- Performance tuning Â helps increase the capacity of simultaneous calls within an SMB office
- Outbound caller ID customization Â each phone extension is equipped with caller ID so callbacks are directly routed to individual phone system users, rather than directed back through the main number
- Diagnostic tools Â diagnostic tests are now run online, helping users identify connection problems with VoIP providers, IP phones and/or the mail server, in addition to offering possible remedies
ÂSwitchvox is dedicated to improving IP PBX technology in a way that makes it affordable and easy to deploy, while providing the most advanced features in the industry,Â said Joshua Stephens, CEO of Switchvox. ÂWith this latest version of Switchvox, SMBs can finally afford to deploy a system with features that have been entirely out of reach due to high cost. A feature like geographical call routing that offers the ability to send a sales call originating from the West Coast directly to the rep that handles this regionÂs sales, can really revolutionize a businessÂ communications.Â
Built on open source software, Switchvox is extremely inexpensive when compared to traditional PBX products, with systems starting as low as $995 (MSRP). The latest applications will be included, at no extra cost, in current systems. Existing customers can upgrade for free if they have a Switchvox support contract. The upgraded Switchvox system is available immediately and can be purchased online by visiting http://www.switchvox.com.
Switchvox is a leading provider of PBX and VoIP phone systems for small- to medium-sized businesses (SMBs). The companyÂs flagship product enables SMBs to easily and affordably create and manage their phone system, using traditional analog lines, as well as VoIP services. Based on Linux and other open source software, Switchvox has created software products that fit business and consumer needs. Headquartered in San Diego, California, more information on the company can be found at http://www.switchvox.com.
Digium is the creator and primary developer of Asterisk, the industryÂs first Open Source PBX. Used in combination with DigiumÂs PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, and Ethernet architectures.
Digium solutions reduce the costs of traditional TDM and VoIP implementations through Open Source, standards-based software and next-generation gateways, media servers and application servers. Digium hardware supports traditional voice protocols, including PRI, RBS, FXS, FXO, E&M, Feature Group D, Groundstart, Loopstart and GR-303. Data protocols include PPP, Cisco HDLC, and Frame Relay. For packet voice, Asterisk supports IAXÂ (Inter-Asterisk eXchange), SIP, MGCP, Cisco SkinnyÂ® (SCCP) and H.323 VoIP protocols.
Code for Asterisk, originally written by Mark Spencer of Digium Inc., has been contributed from Open Source software engineers around the world. It supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. It also supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure. Using the Inter-Asterisk eXchange (IAXÂ) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using packet voice, it is possible to send data such as URL addresses and images in-line with voice traffic, allowing advanced integration of information.
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