AX522 - New VoIP / PSTN 2-Line Intelligent Switch Launched

Posts: 4 · New Forum Member Joined: Nov 01, 2007 Posts: 4

Would you like to have one phone serving both your Vonage Voip line and your standard PSTN Line?

Would you like to automatically make outgoing calls on your Voip Line, while allowing incoming calls through your standard PSTN Line.

Well below is the product that many Vonage customers are turning to:-

Cooltronics have just launched of the new AX522 2-Line Intelligent Switch by specialist Voip manufacturer Artech. The AX522 offers a substantial upgrade over the previous and respected AX520 unit. The AX522 offers a complete re-design on its predecessor, offering unparalleled compatibility and stability across all Voip platforms, as well as incorporating many new and enhanced features. If you are suffering from the hassle of having a separate phone for your Voip system, then the AX522 is for you. With a few simple connections, both your PSTN and Voip lines can be combined to a common phone. The AX522 can be configured to receive all incoming calls through the PSTN line and all outgoing calls through the Voip line. The AX522 intelligently monitors both lines for incoming calls. Also the unit can be programmed with pre-set codes to automatically route certain numbers through pre-determined lines. The AX522 also allows you to switch the incoming and outgoing lines with a switch, of force dial through the PSTN line by pressing #0 on your phone. The new AX522 also features polarity switches for both lines, making redundant the need for RJ-11 / BT Cross Over cables. Another smart feature is that the unit can be programmed to switch to different lines based on the dialling code etc.

For more information go to www.cooltronics.co.uk

www.cooltronics.co.uk

Posts: 1 · New Forum Member Joined: Jun 24, 2009 Posts: 1

I am now looking for Voip solutions. And found information about Voip sdk.
According to their website www.voipsipsdk.com
Voip sdk is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, OpenSER other.

They have all features I need:
# Dynamically loadable codecs
# Registrar support
# Play wav files into conversation
# Record conversation into file
# Hold/Retrieve call
# Forward Call (Blind Call Transfer)
# Transfer Call (Attended Transfer)
# Mute Sound
# VPN support
# Noise reduction
# Auto gain
# Jitter buffer parameters
# Samples on Delphi, C#, VB, VB.NET, C++ 2005, C++ 6.0, HTML (SIP ActiveX)
# Windowless samples on C++ and .NET
# DTMF
# Adaptive silence detection
# Adaptive jitter buffer
# STUN support
# Comes as ActiveX control

But before I will download the evaluation version I would like to hear other people experience.