It is not clear how exactly g729 pass-through can be enabled. I have a SIP call off a gateway come into an Asterisk menu, and then I send the SIP call to another SIP gateway using Dial(). Even though codec preferences have g729 listed first, it never gets used.
Without actually seeing your config files it's hard to guess as to why that might be. Also keep in mind that if you answer the call in the dialplan and want to play messages, you will either need those messages already formatted as G.729 files or G.729 encoder licenses to be able to play them to the caller.
Both gateways have separate peer entries in sip.conf, and both have canreinvite=yes set. Can Asterisk change the media type during a re-Invite? The call is answered as g711u initially, and then Asterisk plays a menu, and then does a Dial(). I can see Asterisk doing the reInvite, but the protocol stays at g711u.
No, Asterisk never changes the codec once the call is established, even when redirecting the media elsewhere.
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