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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