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<DIV><FONT size=2>I did a lot of silly things since I learn FS
:D</FONT></DIV>
<DIV><FONT size=2>but not to try to transcode on FS if I use a proxy
media....</FONT></DIV>
<DIV><FONT size=2>I would just to minimize the transcoding....</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> Monday, December 20, 2010 4:29
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] members
audio conference</DIV>
<DIV><BR></DIV><BR><BR>
<DIV class=gmail_quote>On Mon, Dec 20, 2010 at 12:02 PM, 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>> </FONT><FONT size=3>FS core converts every member
from their codec to L16 </FONT></DIV>
<DIV>ok I understand, so is it possible to change from L16 to another
codec in a conference dialplan ?</DIV></DIV></BLOCKQUOTE>
<DIV>I don't think you understood what Steven was saying. You don't need to
worry about the transcoding. As long as there is a codec that FS supports then
the transcoding happens without you having to do anything. Unless you've done
something silly like proxy media then you shouldn't need to worry about
codecs.</DIV>
<DIV><BR></DIV>
<DIV>-MC</DIV>
<DIV><BR></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>
<DIV bgcolor="#ffffff">
<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"><B>From:</B> <A
title=steveayre@gmail.com href="mailto:steveayre@gmail.com"
target=_blank>Steven Ayre</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> Monday, December 20, 2010 2:48
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users]
members audio conference</DIV>
<DIV><BR></DIV>
<DIV>More specifically FS core converts every member from their codec to
L16 which is given to the conference. The conference combines all speaking
channels and the resulting L16 is given back to the core to send to the
members, the core converting to the correct codec for each
member.<BR><BR>Steve on iPhone</DIV>
<DIV><BR>On 20 Dec 2010, at 17:59, Michael Collins <<A
href="mailto:msc@freeswitch.org" target=_blank>msc@freeswitch.org</A>>
wrote:<BR><BR></DIV>
<DIV></DIV>
<BLOCKQUOTE type="cite">
<DIV>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" target=_blank></A><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>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>
<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></A><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>
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