[asterisk-users] asterisk 1.4 chan_h323, help please...

[Jason Kim]

Hi,

My configuration is SipPhone<-->*1<--->*2.
My asterisk version is 1.4beta3.
I installed pwlib,openh323,chan_h323.
When i call from
SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
there is no audio. 
Using 'rtp debug', I can see that rtp packets are
being received.
Rtp packets are being exchanged.
I also tested chan_ooh323, but to fail.
Can anyone recommand best h323 channel driver?

Regards,
Jason.

#------h323.conf for both------------------------
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
context=default

#------dial plan of asterisk1--------------------
exten => *59,1,Wait(1)
exten => *59,2,Dial(H323/3500 at 192.168.1.150)

#------dial plan of asterisk2--------------------
exten => 3500,1,Playback(hello)
exten => 3500,2,Hangup()

#------console messages with 'rtp debug'---------
    -- Executing [*59 at from-internal:3]
Dial("SIP/3503-0921cb88", "H323/3500 at 192.168.1.150")
in new stack
    -- Requested transfer capability: 0x00 - SPEECH
 -- Making call to 3500 at 192.168.1.150:1720 without
gatekeeper.
        == New H.323 Connection created.
        -- root is calling host
3500 at 192.168.1.150:1720
        -- Call token is ip$localhost/29426
        -- Call reference is 29426
        -- DTMF Payload is [pt=101]
    -- Called 3500 at 192.168.1.150
Setting capabilities to 0x8 (alaw)
Capabilities in preference order is (alaw)
Allowed Codecs:
         Table:
   G.711-ALaw-64k <1>
   UserInput/hookflash <2>
   UserInput/RFC2833 <3>
   UserInput/dtmf <4>
 Set:
   0:
     0:
       G.711-ALaw-64k <1>
     1:
       UserInput/hookflash <2>
     2:
       UserInput/RFC2833 <3>
       UserInput/dtmf <4>

        -- Sending SETUP message
        -- Transmitting RFC2833 on payload 101
        -- Started logical channel: receiving
G.711-ALaw-64k
                -- channelsOpen = 1
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 127.0.0.1
                -- remotePort: 13710
                -- ExternalIpAddress: 192.168.1.116
                -- ExternalPort: 29388
        -- Started logical channel: sending
G.711-ALaw-64k
                -- channelsOpen = 2
                External RTP Session Starting
                RTP channel id 1 parameters:
                -- remoteIpAddress: 127.0.0.1
                -- remotePort: 13710
                -- ExternalIpAddress: 192.168.1.116
                -- ExternalPort: 29388
        - Progress Indicator: 8
    -- H323/192.168.1.150-3 is making progress passing
it to SIP/3503-0921cb88
        -- Inbound RFC2833 on payload [pt=101]
Peer capability is G.711-ALaw-64k <1>
Found peer capability G.711-ALaw-64k <1>, Asterisk
code is 8, frame size (in ms) is 20
Peer capability is UserInput/hookflash <2>
Peer capability is UserInput/RFC2833 <3>
Peer capability is UserInput/dtmf <4>
Peer capabilities = 0x8 (alaw), ordered list is (alaw)
        =-= In OnConnectionEstablished for call 29426
                -- Connection Established with "3500"
    -- H323/192.168.1.150-3 answered SIP/3503-0921cb88
        -- Received Facility message... 
Got  RTP packet from    192.168.1.204:16434 (type 00,
seq 014405, ts 328224084, len 000240)
Sent RTP packet to      127.0.0.1:13710 (type 08, seq
008392, ts 000096, len 000160)
Got  RTP packet from    192.168.1.204:16434 (type 00,
seq 014406, ts 328224324, len 000240)


 
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