Well, PCMA and PCMU are included. Note this line:

m=audio 17262 RTP/AVP 8 0 101 13

8 = PCMA, 0 = PCMU

Not sure why there's no GSM in there but if you review the entire debug log for the call in question it may yield a clue.

FYI, IIRC this variable will give you what you are looking for on the extra SDP lines:

https://wiki.freeswitch.org/wiki/Channel_Variables#verbose_sdp

-MC


On Thu, Jun 5, 2014 at 10:48 AM, Ričardas Stoma <ricardas.stoma@gmail.com> wrote:
I have created the following extension:

<extension name="my_test">
  <condition>
    <action application="bridge" data="{absolute_codec_string='PCMA,PCMU,GSM'}sofia/external/1001@192.168.1.78"/>
    <action application="hangup"/>
  </condition>
</extension>

then i call to this extension, freeswitch dials to 1001@192.168.1.78 and when i check the content of SDP, included in a SIP packet,  i see something like this:

v=0
o=FreeSWITCH 1401972565 1401972566 IN IP4 192.168.1.65
s=FreeSWITCH
c=IN IP4 192.168.1.65
t=0 0
m=audio 17262 RTP/AVP 8 0 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

why my codecs from 'absolute_codec_string' are not included in SDP? I would expect to see these lines included:

a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000


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