no, I'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.
D-

----- Original Message -----
From: "Anthony Minessale" <anthony.minessale@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording and outbound rtp

is it during a bridged call?

On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users@digitaldan.com> wrote:

Hi,

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.

The problem I'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'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-

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