On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos@madovsky.org> wrote:
> 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 ?
I don't think you understood what Steven was saying. You don't need to worry about the transcoding. As long as there is a codec that FS supports then the transcoding happens without you having to do anything. Unless you've done something silly like proxy media then you shouldn't need to worry about codecs.

-MC

----- 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@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@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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