http://wiki.freeswitch.org/wiki/Variable_record_waste_resources On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical <technical@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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- -Rupa