Yes, I am hoping that this Cisco supports SPAN (Switched Port Analyzer) because that is exactly what I need. Unfortunately, that is a job for someone else involved in this project due to political and business reasons, so I can not implement it or even check if it has that feature.<br>
<br>The problem with eavesdrop and mod_spy is that they depend on the listener/recorder initiating the dialplan action. The recorder is passive so the stream must be sent to it as another party to the call, unless I am missing something here.<br>
<br>Thanks.<br><br>Bote<br><br><br><div class="gmail_quote">On Wed, Apr 4, 2012 at 7:20 AM, Saugort Dario Garcia Tovar <span dir="ltr"><<a href="mailto:dgarcia@anew.com.ve">dgarcia@anew.com.ve</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
You could do this doing it at router level. This kind of recorders
just capture rtp streams but not get involved in the call flow. So
if you router have the feature, just pick a port and reproduce all
traffic in that port then connect it to your recorder. In that way,
your recorder will see all rtp packet in your netwokr<div><div class="h5"><br>
<br>
<br>
On 4/4/2012 2:02 AM, Bote Man wrote:
</div></div><blockquote type="cite"><div><div class="h5">It's a ridiculously expensive stand-alone recording
system that can record RTP streams. It's not up to me to change it
at this point.<br>
<br>
My leg C notion is a fallback in case it doesn't grok SIP calls as
advertised; everything else in this project has fallen short of
expectations (except FreeSWITCH) so I expect this to fall short as
well. :- /<br>
<br>
Bote<br>
<br>
<br>
<br>
<div class="gmail_quote">On Wed, Apr 4, 2012 at 2:05 AM, Avi
Marcus <span dir="ltr"><<a href="mailto:avi@avimarcus.net" target="_blank">avi@avimarcus.net</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Use a media bug to record the call?
<div>API: <a href="http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record" target="_blank">http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record</a></div>
<div>Dialplan: <a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session" target="_blank">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session</a></div>
<div><br>
</div>
<div>Or did you mean something completely different?</div>
<div><br clear="all">
<div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br>
<br>
<div class="gmail_quote">
<div>
<div>On Wed, Apr 4, 2012 at 7:03 AM, Bote
Man <span dir="ltr"><<a href="mailto:bote_radio@botecomm.com" target="_blank">bote_radio@botecomm.com</a>></span>
wrote:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div> What are the ways that I can set up
an outbound call to send an additional leg to an
external logging recorder?<br>
<br>
Right now I'm trying to use mod_conference, but it
just ends the call as soon as it begins, even
though I have another dialplan extension that sets
up several endpoints in a conference bridge just
fine.<br>
<br>
I thought about using eavesdrop, but that is
initiated by and sends audio to leg A. Plus it
joins an existing call whereas in this application
I'm establishing a new call, so that won't work. I
need to send the audio to leg C from the start.<br>
<br>
I'm hoping I don't have to write a script, because
I'm on the customer's site and I'm running out of
time.<br>
<br>
Any ideas? Thanks.<span><font color="#888888"><br>
<br>
Bote<br>
</font></span><br>
</div>
</div>
</blockquote>
</div>
</div>
</div>
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<div>-- <br>
Atentamente,
<br>
<b>Dario García</b>
<br>
Consultor.
<br>
<br>
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