We would have to code in a feature to purposely write silence back during a recording that does not currently exist.<br>You could perhaps post it on the bounty section in jira.<br><br><br><div class="gmail_quote">On Tue, Feb 24, 2009 at 2:02 PM,  <span dir="ltr">&lt;<a href="mailto:freeswitch-users@digitaldan.com">freeswitch-users@digitaldan.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div style="font-family: Times New Roman; font-size: 12pt; color: rgb(0, 0, 0);">no, I&#39;m matching the incoming sip call via the destination number in my public context and executing the javascript appliaction.  This app directly answers the call and records it until the user hangs up.<br>
D-<div><div></div><div class="Wj3C7c"><br><br>----- Original Message -----<br>From: &quot;Anthony Minessale&quot; &lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>
Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain<br>Subject: Re: [Freeswitch-users] Recording and outbound rtp<br><br>is it during a bridged call?<br><br><br><div class="gmail_quote">On Tue, Feb 24, 2009 at 11:49 AM, Dan <span dir="ltr">&lt;<a href="mailto:freeswitch-users@digitaldan.com" target="_blank">freeswitch-users@digitaldan.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div style="font-family: Times New Roman; font-size: 12pt; color: rgb(0, 0, 0);"><div style="font-family: Times New Roman; font-size: 12pt; color: rgb(0, 0, 0);">

Hi,<br><br>I
have a small javascript application that accepts a call, retrieves some
dtmf digits and then records the call to an icecast server. This works
great. <br><br>The problem I&#39;m having is that when the call is being
recorded freeswitch is no longer sending rtp packets back to the
originating caller, in my case a Cisco 5300 with a bunch of  T1 voice
circuits in it.  This makes sense, since no voice data back is being
generated.  Unfortunately my Cisco gear has rtp inactivity timers set
up to hang up a call after 3 minutes of no incoming rtp packets, this
is a global setting that cannot be configured for a single dial peer. 
Does anyone have a suggestion to generate rtp packets every once in a
while?  I tried setting comfort noise which did not seem to send
anything.  I could try playing a empty/short wav file every minute or
so but the javascript call session.record is blocking, would a
traditional javascript timer and callback to play a wav file be my best
bet or is there a better approach? I&#39;m using FreeSWITCH Version
1.0.trunk (12108M) on debian etch.<br><br>Thanks!<br><font color="#888888">Dan- <br></font></div></div></div><br>_______________________________________________<br>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>

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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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