[Freeswitch-users] RTP/RTCP media whilst recording

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