[Asterisk-Users] Transcoding

[Rich Adamson]

> Why didn't I think of using that command...
> 
> It shows all "-" for G729a which is presumably why I'm having a problem

That would be a problem.
 
> I have purchased 20 licenses from Digium, downloaded binary, registered the binary correctly, 
placed it in the correct directory and it is listed specifically in SIP.conf
> 
> I'm sure that I have had some calls between SIP phones using G729a via Asterisk (not 
re-invited)
> 
> How can I be sure that the G729a codec is working correctly?

The easiest way is to define one sip phone with g729 only and another
with ulaw only, and place a call. If you can talk, transcoding is working.
If not, more then likely you have a g729 registration problem. Be sure
canreinvite=no is set correctly on those test extensions.

Watch the CLI for the call setup and if there isn't enough data there,
might try a 'sip debug'.

Once the call is in progress, do a 'sip show channels', identify the
channel in use, then do a 'sip show channel xxxxxx' replacing the xxxx
with the appropriate channel string. Might also be helpful to look at
'sip show peer yyyy' where yyyy is the extension number for one of the
test phones.

FWIW, I rebuilt asterisk on a new box, different OS distro, etc. When
I went to re-register the g729 licenses, I had a bitch of a time getting
it done. Called digium support, they could not see my asterisk doing
any registration even after following their exact syntax to the character.
They logged in remotely, executed the same command (so they said), and
it registered immediately. Seemed verrrry flaky to me, and I've been
around this stuff for a couple of years.