absolute_codec_string not working in dialplan - freeswitch

Question

I got some trouble with using absolute_codec_string param.
My call scenario is pretty simple: caller <--> FS <--> callee.
My caller compose m=audio 7078 RTP/AVP 8 0 101 in its INVITE and I'm doing <action application="bridge" data="{absolute_codec_string=PCMU}sofia/gateway/callee/$1"/> in the dialplan.
I expected FS to use only PCMU m=audio 22952 RTP/AVP 0 101 talking to the callee
But FS still use m=audio 22952 RTP/AVP 8 0 101 in the INVITE to the callee.
Note that when I'm using originate application in fs_cli, things are good.
originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100
then I have FS with proper behavior in transcoding, caller has m=audio 31184 RTP/AVP 0 101 received, and callee has m=audio 21922 RTP/AVP 8 101 received.
Not sure what I'm missing, any help would be appreciated.
Edit
I was using FS media_proxy, so absolute_codec_string won't work, any another way to limit codecs used for callee to fulfill my purpose?
e.g: caller use PCMA,PCMU, I want to cut PCMA out of SDP and offer PCMU only to callee
rgds,
Loi Dang

Answer 1

Guess I have the answer for my own question
Briefly, I modify the whole remote sdp with variable switch_r_sdp instead of dealing with absolute_codec_string
<condition field="${switch_r_sdp}" expression="/(.*)(m=audio \d+ RTP\/AVP)([ \d]+)(.*)/s" break="never">
<action application="set" data="switch_r_sdp=$1$2$ 8 101$4"/>
<!-- Then just call bridge normally -->
</condition>
More info: https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation#CodecNegotiation-Modifyingthecodecwhenusingproxymediamode
rgds,
Loi Dang