http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical <techni...@ttnc.co.uk> wrote:
> Hi There
>
> Our Freeswitch cluster receives inbound calls via a SIP trunk from our
> supplier. I currently have an issue where when a call is sent to voicemail
> using session:execute("record"), our supplier will terminate the call with a
> BYE approximately 30 seconds into the recording.
>
> They believe the reason for this is our Freeswitch servers are failing to
> send any RTP/RTCP media while in the recording stage, and therefor they think
> the call is dead.
>
> Is there a way to force Freeswitch to send RTP packets while in the recording
> stage that I'm missing?
>
> Oh, I'm running pretty much the latest svn truck.
>
> Any help appreciated.
>
> Thanks
>
> Russ
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--
-Rupa
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