> 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. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
