Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris