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.<div>-MC<br><br><div class="gmail_quote">On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <span dir="ltr"><<a href="mailto:infos@madovsky.org">infos@madovsky.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div bgcolor="#ffffff">
<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><div></div><div class="h5">
<div style="font:10pt arial">----- Original Message ----- </div>
<div style="font:10pt arial;background:#e4e4e4"><b>From:</b>
<a title="msc@freeswitch.org" href="mailto:msc@freeswitch.org" target="_blank">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" target="_blank">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" target="_blank">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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<br></blockquote></div><br></div>
</div></div><p>
</p><hr><div class="im">
<p></p>_______________________________________________<br>FreeSWITCH-users
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