Hi, I don't have high hopes that this is possible, however I'll give it a shot here :) I have an IVR application which plays a prompt and then hangs up with a BYE. When the BYE is sent, it's intercepted by an opensips b2bua and a reINVITE is issued upstream, followed by a downstream invite (possibly to another IVR or PSTN). The problem I have is that between the BYE from the IVR and the placing of the next call there is often silence, the user will only hear ringing if the downstream INVITE responds by sending RTP to the media IP and port of the caller. My question is, is it possible to have freeswitch/another program somehow continue to send media to the origination IP/port for a few seconds after the BYE has been issued? If it's not possible with freeswitch does anyone know of a third party app which could be invoked to send RTP media to a known IP and port? I know it's a bizarre request but if anyone could point me in a direction I would be grateful. Thanks Pete