[Freeswitch-users] RTP/RTCP media whilst recording

[Rupa Schomaker]

http://wiki.freeswitch.org/wiki/Variable_record_waste_resources

On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical <technical at 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