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<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>I have a problem with incoming
calls being recorded after a supervised transfer.<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>Incoming is CAPI BRI
-> Asterisk -> Supervised Transfer -> SIP.<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>Call comes in, receptionist
answers, caller put on hold, Asterisk Monitor is recording, caller is on Hold,
Callee picks up the call, Asterisk Monitor Stops.<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span class=postbody1><font
size=1 face=Verdana><span style='font-size:9.0pt;font-family:Verdana'>All recorded
calls are named CallerID to Exten. </span></font></span><o:p></o:p></p>
<p class=MsoNormal><font size=1 face=Verdana><span style='font-size:9.0pt;
font-family:Verdana'>Receptionist sees the incoming PSTN callerID, yet when we
get a transfer from the receptionist, we see her SIP callerID, not the incoming
callerID from the PSTN?<br>
<br>
Which rules out, <span class=postbody1>putting a Monitor line into our
macro-stdexten, it will record, but the filename will be local SIP
CallerID's, and we end up with two files for the one call.</span></span></font><span
class=postbody1><o:p></o:p></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>We use Cisco 79XX. <o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>Is there a way to continue the same
recording after a transfer?<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=postbody1><font size=1 face=Verdana><span
style='font-size:9.0pt;font-family:Verdana'>Is there a way to pass on the
Incoming callerID from PSTN to the SIP phones that have the call transferred to?</span></font></span><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>TA.<o:p></o:p></span></font></p>
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