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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.37s
  • Memory usage: 2.18MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 0.71

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

Image

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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