Hi All, I have a querstion about freeswitch conference caller controls. If anyone can give some hint/help, I'm very appreciated. My freeswitch server has been connected to a SIP provider's server (IDT server), and can dial out to normal cellphone numbers. Now I want to start a conference call using "conference <conf-name> dial ..." command, when the destination phones join the conference, the phone callers can't control the conference by pressing the keys specified in <caller-controls> section of conference.conf.xml. (In other words, if I press key '0', it can't mute myself; press '#', it can't hang up, etc. I use the default caller-controls group.) I want to know in this situation, whether the cellphone's key tone can't be transferred to freeswitch server? Can freeswitch server only receive DTMF encapsulated in RTP packet transferred over IP? I can control the conference when I use X-Lite as client, and freeswitch outputs following log when I press key '0' in X-Lite interface: ================================== freeswitch@eli-desktop> 2012-02-29 17:02:59.349272 [DEBUG] switch_rtp.c:2428 RTP RECV DTMF 0:960 2012-02-29 17:02:59.367880 [DEBUG] mod_conference.c:2919 Queueing file '/usr/local/freeswitch/sounds/en/us/callie/conference/conf-muted.wav' for play ================================= In the case of cellphone, when users press key, the DTMF is generated and transferred via PSTN channel to the IDT server, is this correct? and if so, is it up to IDT server to encapsulate the DTMF in RTP packets and send it to freeswitch server? -- Best Regards Erjian