[Freeswitch-users] absolute_codec_string not working

[Michael Jerris]

using proxy_media is my best guess but can’t tell with this little info.

> On Nov 22, 2016, at 5:27 AM, Lợi Đặng <loi.dangthanh at gmail.com> wrote:
> 
> 
> 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 <mailto:loi.dangthanh at gmail.com>
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