more weird, if I choose a codec from the sip phone based on 8000hz it doesn't work. it's like a rate conversion has to be forced to make the audio work on the conference bridge... ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Sunday, January 16, 2011 12:10 PM Subject: Re: [Freeswitch-users] bridge a call to remote conference after some test I confirm that it should force to offer a codec at the same rate of the conference, example <action application="bridge" data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/> as GSM is a 8000hz code so it works now... thanks ----- Original Message ----- From: Steven Ayre To: FreeSWITCH Users Help Sent: Sunday, January 16, 2011 4:20 AM Subject: Re: [Freeswitch-users] bridge a call to remote conference Try answering the call before you enter the conference, possibly with a small sleep inbetween too. That might give the media time to startup before the conference IVR starts. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep -Steve On 16 January 2011 03:54, Madovsky <infos@madovsky.org> wrote: everyhting is ok unless that conference ivr can't be heard (like fraction of sec of sound every 2/3 sec) but the hold music (in 8000hz) works well. should I set any special codec in the legB conference bridge ? 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