[Freeswitch-users] absolute_codec_string not working

[Lợi Đặng]

Hi List, 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,GSM}sofia/gateway/callee/$1"/>` in the
dialplan.
But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the
callee.
not sure what I'm missing, helps would be appreciated.

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`.
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.

rgds,
Loi Dang Thanh
Phone : 84.1224.735.448
Email : loi.dangthanh at gmail.com
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