It is still the same without single quote marks around PCMA,PCMU,GSM...

But i think everything is OK with my SDP, i just made a call to my asterisk server and i see that asterisk receives correct codecs:

Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw)

So i guess i had to learn more about SDP and codecs before posting my issue, sorry guys :-)


2014-06-06 15:47 GMT+03:00 Anonio Silva <asilva@wirelessmundi.com>:
Can you try:

{absolute_codec_string=PCMA,PCMU,GSM}   ===> without '



On Fri, 2014-06-06 at 11:36 +0300, Ričardas Stoma 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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org