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
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