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@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 -----
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@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@16000k"/>
 
but no success
 
Thanks

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