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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
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AU-600

The AU-600 is a USB device which provides a traditional PSTN FXO and FXS interface to a Windows machine running the Skype VoIP client software. It allows users to use a regular analog telephone as the audio device on their Skype client, as well as rudimentary call routing between the Skype network and the PSTN. Some users have also successfully used the AU-600 in concert with an FXO port on an Asterisk server as a single-channel gateway between their Asterisk call routing and the Skype network (A sort of hardware chan_skype). They are available for purchase online from a seemingly endless number of VoIP vendors for around USD$50.

Image

Features:

  • Continue to make and receive regular calls as you normally do
  • Make and receive Skypeâ„¢ calls using your standard telephone
  • Forward Skypeâ„¢ calls to your mobile phone
  • Make Skypeâ„¢ calls from your mobile phone even when you are away from your computer
  • Switch between a Skypeâ„¢ call and a regular phone call

Requirements:

  • Windows 2000 or XP
  • Available USB port (1.0, 1.1 or 2.0)
  • Skypeâ„¢ version 1.1 or higher (www.skype.com) must be installed
  • 128MB RAM
  • 10MB available hard disk space

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Contact :


ATCOM technology co.,limited
Address Rm.6A/B,Shangtian building,No.70,Nanyuan Rd., Shenzhen,China
Tel (86)+755-83018469 83018569 83018769
Fax (86)+755-83018319
http://www.atcom.cn
E-mail sales@atcom.com.cn
Support Forum: bbs.pa1688.com

See also:


Created by petersun, Last modification by Paul Kelly on Mon 27 of Nov, 2006 [17:42 UTC]

Comments Filter

by jklaas on Monday 21 of August, 2006 [15:23:14 UTC]
This appears to be based on (OEM'd from) SmartLink. There is an au600 driver being worked on over in the alsa project (http://alsa.opensrc.org/au600). There's more information there on this device.

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