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<DIV><FONT size=2>nohgint strange on logs.</FONT></DIV>
<DIV><FONT size=2>but I guess it's a codec and rate problem.</FONT></DIV>
<DIV><FONT size=2>tried with different SIP phones and it works.</FONT></DIV>
<DIV><FONT size=2>how a conference manage the codecs ?</FONT></DIV>
<DIV><FONT size=2>I know the rate can be set in profile,</FONT></DIV>
<DIV><FONT size=2>but how conference codec is managed</FONT></DIV>
<DIV><FONT size=2> </FONT><FONT size=2>if all members have different codec
and rate ?</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=msc@freeswitch.org href="mailto:msc@freeswitch.org">Michael
Collins</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">FreeSWITCH Users Help</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, December 17, 2010 7:54
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] members
audio conference</DIV>
<DIV><BR></DIV>What do you see in the debug logs? Did you compare the logs for
a working vs. non-working call? Anything different?
<DIV><BR></DIV>
<DIV>-MC<BR><BR>
<DIV class=gmail_quote>On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <SPAN
dir=ltr><<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>></SPAN> wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2>when the first member creates and enters in a new
conference</FONT></DIV>
<DIV><FONT size=2>everything is ok. but if a new memeber enters there is no
audio</FONT></DIV>
<DIV><FONT size=2>in the conference, unless the ivr.</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>I have a very simple conference dialplan like
this</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2> <extension
name="create_conference"><BR>
<condition field="${sip_to_uri}"
expression="^000(\d{10,15})@$${domain}$"><BR>
<action application="set"
data="instant_ringback=true"/><BR>
<action
application="ring_ready"/><BR>
<action application="sleep"
data="3000"/><BR>
<action
application="answer"/><BR>
<action application="conference" data="<A
href="mailto:$1-$%7Bdomain_name%7D@wideband%22/"
target=_blank>$1-${domain_name}@wideband"/</A>><BR>
</condition><BR>
</extension><BR></FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>I tried to add</FONT></DIV>
<DIV><FONT size=2><action application="export" data="<A
href="mailto:absolute_codec_string=speex@16000k%22/"
target=_blank>absolute_codec_string=speex@16000k"/</A>></FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>but no success</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT
size=2>Thanks</FONT></DIV></DIV><BR>_______________________________________________<BR>FreeSWITCH-users
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<P>
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<P></P>_______________________________________________<BR>FreeSWITCH-users
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