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

[Julio Arruda]

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>