[Asterisk-Users] Leadtek BVA8051 / Sipphone.com CallInOne withAsterisk?

Brandon Patterson siptech at livevoip.com
Wed Dec 8 18:48:28 MST 2004


Leadtek requires a firmware upgrade at the least. Have you done this?
If not email me and I will provide it free to you.

Brandon Patterson
LiveVoip LLC




> Jerry Glomph Black wrote:
> > I have a lot of experience, all of it pretty good, with various Sipura
> > products, Grandstreams,  Zultys, IAXy, and numerous SIP/IAX soft phones
> > connecting into Asterisk as clients.   Good sound quality, great
> > reliability.
> >
> > I've tried two of the units named in the subject line, and frankly I'm
> > frustrated.   Calls usually start out OK, but within a brief period the
> > sound goes totally to Hell.  Sounds like the packets are being
> > reassembled out of order, because there is a regular candence to the
> > garbling.  Problem is almost always on the receiving end, the distant
> > party on the call seems to get OK audio.
> >
> > Most annoying is that when I log the device directly into a VoIP
> > provider (have tried FWD, Stanaphone, and Sipgate.de) IT WORKS FINE!
> >
> > I've tried asterisk boxes on the local LAN, and thousands of miles away.
> > Asterisk versions from 0.7.2  to 1.0.3.
> >
> > Results have been consistently flaky, I've tried flash upgrading, makes
> > no difference.   Have tried all sorts of config tweaks on the phone as
> > to buffer size, etc.
> >
> > Google has almost NO info on these things, they have one nice feature
> > which is easy autoswitching between POTS and SIP calls in both
directions.
> > Any experience or hearsay out there in Asterisk land?
>
> Not very helpful, I know..but..
>  From what I understand, these are based in the same hw/sw as the
> Packet8 DTA310 ? (audacity based gear)
> I use DTA310 for some time with * and seems to work fine for my
> purposes, and with good quality.
> Anyway, you may want to post the configuration you use for the leadtek,
> may give the others a hint ?
> [], <O-O>
>
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