As5350 VOIP h323 g723 codec problem

Question

I setup the VOIP network with H323. All calls are Gatekeeper routed.
When I make a PSTN call either inbound or outbound, as5350 could not
establish a call other than g729 codec.
Question 1. Why AS5350 uses g729 codec as default ? How you setup a
g729 default codec?
Question 2. Why AS5350 cannot use other codec like g723? AS5350
release a call right after setup because it can't talk with g723
codec. Gateway was setup only g723 codec to see if AS5350 can talk.
When I say PSTN call, I mean the call initiated from our h323 gateway
to PSTN number.
Gateway --> AS5350 PRI --> PSTN number (outbound call)
PSTN number --> AS5350 PRI --> H323 gateway (inbound call)
Codec is established in the Internet portion between gateway and
AS5350.
Gateway is non cisco equipment and setup as g729 and g723
capabilities.
AS5350 Configuration only necessary
portion----------------------------------------
voice call send-alert
voice rtp send-recv
voice service pots
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0
h323
voice class codec 1
codec preference 1 g729r8
codec preference 2 g723r63
codec preference 3 g723r53
voice class codec 3
codec preference 1 g723r63
voice class h323 1
call start fast
controller T1 2/0
framing esf
linecode b8zs
pri-group timeslots 1-24
interface Serial2/0:23
no ip address
isdn switch-type primary-ni
isdn incoming-voice modem
isdn T306 30000
isdn T310 60000
isdn send-alerting
isdn sending-complete
no cdp enable
voice-port 2/0:D
echo-cancel coverage 128
no vad
bearer-cap Speech
dial-peer voice 1 pots
preference 1
destination-pattern 1.T
progress_ind setup enable 3
no digit-strip
port 2/0:D
forward-digits all
dial-peer voice 2132332745 voip
destination-pattern 2132332745
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ras
interface FastEthernet0/0
ip address x.x.x.x 255.255.255.0
no ip proxy-arp
duplex auto
speed auto
no cdp enable
h323-gateway voip interface
h323-gateway voip id AAAA ipaddr x.x.x.x 1719
h323-gateway voip h323-id as5350

Answer 1

What solution did you use, if you don't mind me asking?
As far as CUBE goes, I think I have the answer. It can be done only using 2 transcoders.
CUBE can't add or remove transcoder mid-call, and while CUCM can add a transcoder during transfers it seems it can't remove a transcoder mid-call. So in our case during the transfer the xcoder just signals media error and drops the call.
The solution is to use one transcoder on CUBE (CME registered) - this one will always xcode the incoming call from g729 to g711, then there needs to be another xcoder on CUCM - CUCM will use it during transfer to transcode between the CUBE call leg (g711) and phone C call leg (g729).
BTW, I need this functionality in UCCE environment with Outbound Dialer and VoIP connection to PSTN (g729 only).