Hello, I've got a problem with FreeSwitch version 1.4.26. Our setup is like this: We have the phones registered at a SipXecs cluster. Freeswitch is used as a gateway (sbc) between SipXecs and another SIP device (Audiocodes Mediant). We aim to bypass the media at the FreeSwitch. At first this works as expected, the media does not go through the FreeSwitch. However when a phone issues a transfer, the refer is handled by SipXecs. The invite with replaces header is sent to Freeswitch (via SipX). FreeSwitch handles this SIP message correctly, but it puts itself in the RTP-stream. The original channel has the variables bypass_media and bypass_media_after_bridge set to true. It seems that these variables are not checked when the invite with replaces header is handled. Anyone an idea how to fix this? Best regards, Sjoerd. ----- Sjoerd Boomstra | Senior Business Consultant | Telecats bv | KvK Enschede 06069106 | Tel: 053 488 99 07 | Fax: 053 488 99 10 | E‑mail: s.boomstra@telecats.nl |