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<DIV><FONT size=2>I got it thanks,</FONT></DIV>
<DIV><FONT size=2>but do you think it would be more interesting to
reduce</FONT></DIV>
<DIV><FONT size=2>bandwidth and latency between nodes and centralize the
conference on one node only</FONT></DIV>
<DIV><FONT size=2>by transferring the incoming user to the right node
?</FONT></DIV>
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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=jmesquita@freeswitch.org href="mailto:jmesquita@freeswitch.org">Joćo
Mesquita</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> Thursday, January 06, 2011 4:33
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users]
mod_conference with cluster ODBC</DIV>
<DIV><BR></DIV>Ok, let me see if I can get this into your head. (giggles)
<DIV><BR></DIV>
<DIV>A conference means that the audio needs to mixed in together so that all
participants can talk/hear each other, right? If you implement something in C
on mod_conference, you are going to essentially do the same as what an ESL app
does. You _need_ to call in from one server to the other so that you can mix
the audio of all the participants. The real advantage would be the management
API being only one for everything and the challenge is exactly that. How to
mute certain users on a conference that is spanning over 10 servers or deaf
them, etc...</DIV>
<DIV><BR></DIV>
<DIV>A SIP "user" is easier because you don't have to bridge audio from
another server necessarily. Got it?</DIV>
<DIV><BR></DIV>
<DIV>Regards,</DIV>
<DIV>Joćo Mesquita<BR><BR><BR>
<DIV class=gmail_quote>On Thu, Jan 6, 2011 at 6:25 PM, Madovsky <SPAN
dir=ltr><<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>></SPAN> wrote:<BR>
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class=gmail_quote>Rupa,<BR><BR>I don't want bother anyone with this thread
but why not<BR>to manage conference as SIP user ?<BR>if someone from server
A call an other who is registered on server B, so<BR>FS do it automatically,
why not with conference ? Or maybe create a param<BR>in mod_conference that
let the choice of the admin to manage unique name in<BR>all cluster or
not.<BR>like <param name="cluster_unique_name" value="true"/><BR>I
will try to understand the C code to hack something like this...<BR>
<DIV class=im><BR>----- Original Message -----<BR>From: "Rupa Schomaker"
<<A href="mailto:rupa@rupa.com">rupa@rupa.com</A>><BR>To: "FreeSWITCH
Users Help" <<A
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>><BR></DIV>
<DIV>
<DIV></DIV>
<DIV class=h5>Sent: Thursday, January 06, 2011 3:01 PM<BR>Subject: Re:
[Freeswitch-users] mod_conference with cluster ODBC<BR><BR><BR>Yes<BR><BR>On
Thu, Jan 6, 2011 at 1:40 PM, Madovsky <<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>> wrote:<BR>>
in case of you have 8 servers you have to do it for each ?<BR>><BR>>
Thanks<BR>><BR>> ----- Original Message -----<BR>> From: joy
this<BR>> To: FreeSWITCH Users Help<BR>> Sent: Thursday, January 06,
2011 2:51 AM<BR>> Subject: Re: [Freeswitch-users] mod_conference with
cluster ODBC<BR>> It works. Thank you everyone.<BR>><BR>> 2011/1/5
Rupa Schomaker <<A
href="mailto:rupa@rupa.com">rupa@rupa.com</A>><BR>>><BR>>>
Use the api: conference <confname> dial [{dial string<BR>>>
options}]<endpoint_module_name>/<destination>
[<callerid_number><BR>>> [<callerid_name>]]<BR>>> To
initiate the call from within conference A on server 1. Have a<BR>>>
corresponding dialplan entry on server 2 to accept the call and add
it<BR>>> into<BR>>> the conference A on server 2. You've now
bridged the two conferences in<BR>>> the<BR>>> two
servers.<BR>><BR>> ________________________________<BR>><BR>>
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