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@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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org