asterisk users - Aug 2005 - Asterisk on FreeBSD-5.4 RELEASE : H323 audio problem

Dear All. 
 
 
I have installed Asterisk-1.0.6 on FreeBSD-5.4 RELEASE via port, so the
chan_h323.so modules already included.
 
 
When i try the SIP channel, asterisk works fine. In this case, i use the XLite
softphone as client, i can hear the voice transfered through clearly. Asterisk
is good!! :D
 
 
The problem is when i try the H323 channel, the voice cannot be transferred
through. I have already use the NetMeeting and SJPhone for clients when
debugging. The clients can call each other, but when the call answered, there
wasn't any voice. So the problems was with audio, not at call setup.
 
 
In the CLI mode, i have the h.323 debug and h.323 trace enabled, and there is no
problem in call setup.
                                          ^^^^^^^^^^^^         ^^^^^^^^^^^ 
 
When i try the h.323 show codecs .... it is empty!! Is this the cause?? 
                       ^^^^^^^^^^^^^^^^^^^ 
When i try the h.323 show tokens ..... it is empty too!!! Is this also the
cause??
                       ^^^^^^^^^^^^^^^^^^^ 
 
But when i try the show codecs the codecs was showed completely. Looks like the
codecs was fine.
                              ^^^^^^^^^^^^^ 
 
Is this a bug, or my configuration fault??? 
 
I have tried Googling all this week, but there is no answer, so help me please!
:D
 
 
------------------------------------------------------------------------ 
[THIS IS MY SIP.CONF H323.CONF AND EXTENSIONS.CONF] 
------------------------------------------------------------------------ 
 
========= 
sip.conf 
========= 
 
[general] 
port=5060 
bindaddr=0.0.0.0 
context=kpvoip-sip 
 
[ridho] 
type=friend 
host=dynamic 
defaultip=202.152.160.109 
musiconhold=default 
context=kpvoip-sip 
;context=demo 
canreinvite=no 
username=ridho 
secret=ahmad 
callerid="ridho" 
nat=no 
;dtmfmode=rfc2833 
 
[koko] 
type=friend 
host=dynamic 
defaultip=202.152.160.106 
musiconhold=default 
context=kpvoip-sip 
;context=demo 
canreinvite=no 
username=koko 
secret=nkholis 
callerid="koko" 
nat=no 
;dtmfmode=rfc2833 
 
[bambang] 
type=friend 
host=dynamic 
defaultip=202.152.160.99 
musiconhold=default 
context=kpvoip-sip 
canreinvite=no 
username=bambang 
secret=bambang 
callerid="bambang" 
nat=no 
 
 
========= 
h323.conf 
========= 
 
[general] 
port = 1720 
bindaddr=202.152.160.108 
context=kpvoip-h323 
 
;disallow=all 
allow=all 
allow=gsm 
allow=ulaw 
alow=alaw 
allow=g723.1 
 
gatekeeper=202.152.160.5 
AlloGKRouted=yes 
dtmfmode=inband 
 
[demo] 
type=h323 
e164=3000 
context=demo 
 
[Ridho S] 
type=friend 
host=202.152.160.109 
;context=default 
context=kpvoip-h323 
 
[cak koko] 
type=friend 
host=202.152.160.106 
;context=default 
context=kpvoip-h323 
 
 
[bambang] 
type=friend 
host=202.152.160.107 
context=kpvoip-h323 
 
 
============= 
extensions.conf 
============= 
 
[kpvoip-sip] 
exten => 1000,1,Dial(SIP/koko) 
exten => 2000,1,Dial(SIP/ridho) 
;exten => 3000,1,Dial(SIP/bambang) 
exten => 3000,1,Dial(H323/202.152.160.109) 
 
[kpvoip-h323] 
exten => 5000,1,Dial(H323/202.152.160.106) 
exten => 4000,1,Dial(H323/202.152.160.109) 
exten => 6000,1,Dial(H323/202.152.160.107) 
 
[demo] 
exten => 2,1,BackGround(demo-moreinfo) 
exten => 2,2,Goto(s,6) 
 
[default] 
;exten => koko,1,Dial(H323/koko,t,20) 
;exten => koko,1,Answer 
;exten => koko,2,Playback,current-time 
;exten => ridho,1,Dial(H323/ridho,t,20) 
exten => 5000,1,Dial,H323/202.152.160.106 
exten => 4000,1,Dial,H323/202.152.160.109 
  
 
------------------------------------------------------------------------------------------------
 
 
 
Best Regards 
Sigit Priyanggoro 
ComLabs Research Group ITB 
http://sigit.no-ip.org