asterisk users - Dec 2004 - Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk?

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?

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>