The call will fail. Asterisk doesn’t know that a G.729 decoder/encoder isn’t available until it actually goes to use one… meaning when the RTP stream begins. This happens after the call is set up at the SIP level, so the session will initiate properly, but no audio will flow.
If recording is going to be a big component of your application and you don’t want to purchase extra licenses, there is a workaround you could implement. If you dump both channels into a MeetMe conference you can record the conference without requiring an extra G.729 license. MeetMe transcodes everything to s-linear, so the recording doesn’t require a license to write a WAV file.