[Freeswitch-users] RTP/RTCP media whilst recording

TTNC - Technical technical at ttnc.co.uk
Wed Dec 23 06:26:40 PST 2009

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. 



More information about the FreeSWITCH-users mailing list