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Mediatrix Offers Support for the Asterisk PBX Telephone System


The versatility of the Mediatrix Analog VoIP Gateways and their seamless integration with a variety of multi-protocol PBX solutions is key to the products continued success



Montreal, QUEBEC, March 14, 2006 – Mediatrix Telecom Inc., the leader in Voice over Internet Protocol (VoIP) access devices and gateways, announced today that its line of Analog Access Devices and Gateways support the Asterisk PBX Telephone System.

The Asterisk open source PBX solution provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It also has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).

The following features are supported by both Asterisk and Mediatrix Analog Units:

RFC 2833, SIP INFO, and inband DTMF transports
SIP Authentication
Blind and Supervised Call Transfer
Call Forward On Busy/On No Answer/Unconditional
Call Waiting
Voice Mail
Conference
Music on Hold
Caller ID
PSTN Trunk Calls (via Mediatrix 1204)
G.711, G.723.1 and G.726 codecs
Fax transmission
IVR

"Asterisk is a proven IP-PBX solution operating in hundreds of organizations worldwide, from small and medium sized installations to organizations with several thousand users," explained Helene Chartier, Vice President Marketing and PLM for Mediatrix. "The of use of an open standards PBX solution like Asterisk in combination with our line of award winning VoIP access devices and gateways, ensures a cost-effective solution that is highly efficient, flexible and secure for customers”

www.mediatrix.ru / www.mediatrix.com