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<DIV><FONT face=Arial size=2>We had a question:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>> With which application and API is possible to
keep full content of RTP audiostream to a file?<BR>> Our FreeSWITCH is
configured in such manner that RTP-traffic directing throw FreeSWITCH,<BR>>
but not between UAs.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>And took answer on it:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>> <A
href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session</A></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>-------------------------------------------------------------------</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>On it HELP we have mase
such extension :</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><extension name="ext-record"><BR>
<condition field="destination_number"
expression="^4010$"><BR> <action application="set"
data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number}
${strftime(%Y-%m-%d %H:%M)}"/><BR> <action
application="set" data="RECORD_COPYRIGHT=(c) 2008, Altron Inc."
/><BR> <action application="set"
data="RECORD_SOFTWARE=FreeSwitch" /><BR> <action
application="set" data="RECORD_ARTIST=Registration" /><BR>
<action application="set" data="RECORD_COMMENT=!!!"
/><BR> <action application="set"
data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/><BR>
<action application="set"
data="RECORD_STEREO=true"/><BR> <action
application="record_session"
data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/><BR>
<action application="set"
data="ringback=${us-ring}"/><BR> <action
application="bridge" data="<A
href="mailto:sofia/nat/${dialed_ext}@localhost:5062">sofia/nat/${dialed_ext}@localhost:5062</A>"
/><BR> <action application="answer" /><BR>
</condition><BR></extension></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>This draft works already, but there
are new questions:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>1. It's seems, that record of RTP session can
occur only when RTP stream is routed through FS: UA-> FS-> UA,
instead of directly (in LAN) UA-> UA.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>For this purpose, as we understand, it is necessary
to set values in variables of FreeSWITCH:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><action application="set"
data="bypass_media=..."/><BR><action application="set"
data="proxy_media=..."/><BR><action application="set"
data="inbound_late_negotiation=..."/> ( for SIP profile nat for
extension ext-record).</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>What values need to be set in variables that
all RTP traffic passed through FreeSWITCH?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>2. At extension ext-record we set record_session
then we do bridge to leg B. <FONT face=Arial size=2>As it is possible
to</FONT> configurate registration from another extension, for
example:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2><extension name="XXXX"><BR>
<condition field="destination_number"
expression="^XXXX$"><BR>
...<BR> <action application="execute_extension"
data="ext-record XML public "/> <BR>
...<BR> <action application="bridge"
data="sofia/nat/YYYY" /><BR> ...<BR>
</condition><BR></extension></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>- where own extension not call any leg B, but only
record session XXXX->YYYY.<BR>How we are able to do this???</DIV>
<DIV> </DIV>
<DIV>Thankful in advance,
Evgeniy.<BR>
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