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 ----- *From:* Michael Collins <msc@freeswitch.org> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *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"/ <$1-$%7Bdomain_name%7D@wideband%22/>> </condition> </extension>
I tried to add <action application="export" data="absolute_codec_string=speex@16000k"/<absolute_codec_string=speex@16000k%22/>
but no success
Thanks
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