[asterisk-dev] Digium G.729 codec binaries updated (or DEA is a bonehead)

[Rob Fugina]

On 9/7/06, Kevin P. Fleming <kpfleming at digium.com> wrote:
>
> ----- Dan Austin <Dan_Austin at Phoenix.com> wrote:
> > I've had a chance to run a few more tests.  I disabled transcode via
> > SLIN and tried a few more calls.  Any transcoding between endpoints
> > (G729<->ULAW) results in a load white-noise like static with no
> > discernable traces of the original audio signal.  A G729 call into
> > a MeetMe room results in an instant segfault.
>
> Very bizarre indeed. I had a couple of people test these before uploading
> them, and they appeared to work properly.
>
> Can you tell which CPU-optimized version you are using, and on what
> specific type of CPU? Does the problem occur if you use a less-optimized
> version (like the i686 or i386 one)?
>
>
I'm having the same issue with g729<->ulaw, but I haven't done anything with
meetme.  I'm using the i686/Ast1.4 version of the codec.  This is my first
experience with g729, as I only bought my first 2 licenses last night, so
I'll be the first to admit there may be something else I'm doing wrong...
I'm using asterisk svn trunk as of last night, also.

Rob
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