[Freeswitch-users] members audio conference

[Madovsky]

> FS core converts every member from their codec to L16 
ok I understand, so is it possible to change from L16 to another codec in a conference dialplan ?
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, December 20, 2010 2:48 PM
  Subject: Re: [Freeswitch-users] members audio conference


  More specifically FS core converts every member from their codec to L16 which is given to the conference. The conference combines all speaking channels and the resulting L16 is given back to the core to send to the members, the core converting to the correct codec for each member.

  Steve on iPhone

  On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:


    The conference itself doesn't do any codec stuff - FreeSWITCH core does. All I can say is compare the working versus non-working logs and look for clues.
    -MC


    On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:

      nohgint strange on logs.
      but I guess it's a codec and rate problem.
      tried with different SIP phones and it works.
      how a conference manage the codecs  ?
      I know the rate can be set in profile,
      but how conference codec is managed
       if all members have different codec and rate ?



        ----- Original Message ----- 
        From: Michael Collins 
        To: FreeSWITCH Users Help 
        Sent: Friday, December 17, 2010 7:54 PM
        Subject: Re: [Freeswitch-users] members audio conference


        What do you see in the debug logs? Did you compare the logs for a working vs. non-working call? Anything different? 


        -MC


        On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:

          when the first member creates and enters in a new conference
          everything is ok. but if a new memeber enters there is no audio
          in the conference, unless the ivr.

          I have a very simple conference dialplan like this

                  <extension name="create_conference">
                          <condition field="${sip_to_uri}" expression="^000(\d{10,15})@$${domain}$">
                                  <action application="set" data="instant_ringback=true"/>
                                  <action application="ring_ready"/>
                                  <action application="sleep" data="3000"/>
                                  <action application="answer"/>
                                  <action application="conference" data="$1-${domain_name}@wideband"/>
                          </condition>
                  </extension>


          I tried to add
          <action application="export" data="absolute_codec_string=speex at 16000k"/>

          but no success

          Thanks

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