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<DIV><FONT size=2>more weird,</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>if I choose a codec from the sip phone based on 8000hz it
doesn't work.</FONT></DIV>
<DIV><FONT size=2>it's like a rate conversion has to be forced to make the audio
work on the conference bridge...</FONT></DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=infos@madovsky.org href="mailto:infos@madovsky.org">Madovsky</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> Sunday, January 16, 2011 12:10
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] bridge a
call to remote conference</DIV>
<DIV><BR></DIV>
<DIV><FONT size=2>after some test I confirm that it should force to offer a
codec</FONT></DIV>
<DIV><FONT size=2>at the same rate of the conference, example</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2><action application="bridge"
data="{absolute_codec_string=GSM,ignore_early_media=true}sofia/internal/${conference}"/></FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>as GSM is a 8000hz code so it works now...</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT size=2>thanks</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=steveayre@gmail.com href="mailto:steveayre@gmail.com">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">FreeSWITCH Users
Help</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, January 16, 2011 4:20
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] bridge
a call to remote conference</DIV>
<DIV><BR></DIV>Try answering the call before you enter the conference,
possibly with a small sleep inbetween too. That might give the media time to
startup before the conference IVR starts.<BR><BR><A
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_answer</A><BR><A
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sleep</A><BR><BR>-Steve<BR><BR><BR>
<DIV class=gmail_quote>On 16 January 2011 03:54, Madovsky <SPAN
dir=ltr><<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>></SPAN>
wrote:<BR>
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<DIV bgcolor="#ffffff">
<DIV><FONT size=2>everyhting is ok unless that conference ivr can't be
heard (like fraction of sec of sound every 2/3 sec)</FONT></DIV>
<DIV><FONT size=2>but the hold music (in 8000hz) works well.</FONT></DIV>
<DIV><FONT size=2>should I set any special codec in the legB conference
bridge ?</FONT></DIV>
<DIV><FONT size=2></FONT> </DIV>
<DIV><FONT
size=2>Thanks</FONT></DIV></DIV><BR>_______________________________________________<BR>FreeSWITCH-users
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