Hi Transfers from local to local user work without problem, but when a person call in via sip provider (selects call type via ivr and he is put into fifo) and somebody answers this call, he is unable to transfer this call to another person (via *1/*3/*4). When he presses *, he is disconnected. Any ideas why ? 2011-08-19 14:55:17.475054 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF *:800 2011-08-19 14:55:17.475054 [DEBUG] switch_channel.c:2775 (sofia/external/693xxxxxx@trunk.dialog.pl) Callstate Change ACTIVE -> HANGUP -- pozdrawiam - Tomasz Hyziak