[asterisk-embedded] AA50 & codec_g729a.so?

[Jason Parker]

Dayton Turner wrote:
> Has anyone compiled codec_g729a.so for the AA50 yet? Im trying to do  
> some load testing on mine running even in passhrough mode, but when  
> using originate/call files, it wants to transcode to slin for the  
> first leg of the call, which fails.  I know g729 is a bit of a  
> challenge with no MMU/FPU, but from what I've read it should be  
> possible. It would also be nice to see this for small tasks like  
> Voicemail -> Email WAV transcoding.
> 
> Alternatively, if anyone has any suggestions on how I might test  
> calling capacity of the AA50 in g729 passthrough mode without having  
> to have a phone place each call, ie generating the calls from the AA50  
> in g729 to an external asterisk box which also speaks g729 and just  
> plays audio back, I'm all ears...
> 
> Dayton

You can use SIPP on another box to generate the calls to the AA50.  On some
Asterisk box somewhere, call in with a phone using g729 and use Record() to
create a g729 audio file.  Then you can just create a SIPP scenario file for a
g729 call (it's pretty simple - and you can probably easily google up an
existing scenario)

-- 
Jason Parker
Digium