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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
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Leadtek

Leadtek is an OEM manufacturer for video phones and SIP ATAs.
http://www.leadtek.com/

Also See This Leadtek Page


Support:



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]

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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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