Hi All,
Firstly, I want to give my thanks to every one in the mailing list,
especially the ones who helped me on my questions. My two questions about
freeswitch has been solved due to the helps provided here.
Today I meet a new question about freeswitch:
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