[Freeswitch-users] members audio conference

[Madovsky]

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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