[Freeswitch-users] Recording and outbound rtp

[Anthony Minessale]

is it during a bridged call?


On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users at 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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>


-- 
Anthony Minessale II

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