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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.

Also See This Leadtek Page

Support:

Leadtek Support Forum

Their video phone products are sold by:

Their SIP ATA models are: Leadtek BVA8051S, BVA8052D, BVA8053R, BVA8055
The BVA8051S ATA is designed to combine your PSTN and SIP calls and send both to a connected phone.
Long distance calls are automatically made via SIP and local calls via PSTN.
SIPphone is current selling it as the Call-in-One for $89 (as of 3/27/04) SIPphone Call-in-One Information

Created by jht2, Last modification by Eric Chamberlain on Tue 27 of Feb, 2007 [19:51 UTC]

I have one of these, beware

by Anonymous on Sunday 12 of September, 2004 [03:28:34 UTC]

It's a great idea on paper, but the SIP stack is flaky as hell. I have one of these, a SIPURA 2000, a Zultys Zip2, and one of the usual Grandstream phones, all on the same net, making calls the same way. The Plug-n-dial loses synch with the data stream regularly, and almost never resynchs. I've tried all manner of adjustments, etc. All of the other devices do very well with the various VoIP services I connect to. Avoid this POS, it's been frustrating.

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