Hey Everyone, I'm using an Audiocodes MP-118 VoIP Gateway. When making a call through the device to a Freeswitch gateway on the other side, it sends the call to my desk phone just fine. When I mute the call, we get the hold music as per the dial plan, everything is good. Then when I unmute... the hold music stops, and... nothing. The call doesn't drop, it just sits there. In the Audiocodes log, we see: 5d:1h:25m:32s ( lgr_psbrdex)(28879 ) recv <-- acEV_BROKEN_CONNECTION, Ch:0 5d:1h:25m:32s ( lgr_flow)(28880 ) #0:RTP_BROKEN_CONNECTION_EV 5d:1h:25m:32s ( lgr_flow)(28881 ) | #0:RTP_BROKEN_CONNECTION_EV I know this isn't entirely a Freeswitch question, but I figure the crowd here might know the answer or where I should look for one. Calls through our Grandstream device work fine, so I think this can be narrowed down to the Audiocodes device. But maybe there's a different codec or option or something I could try to see if I can make it happy? Thanks, -Mike Kallies