[Asterisk-Users] Failure in RTP streaming

[kiel hedjam]

hi,

I use the oh323 driver to answer H323 calls.
The connection is set up normally.

In my extensions.conf file I use:

exten => s,1,Answer
exten => s,2,Playback(demo-instruct)
exten => s,3,Hangup


So that when a call is answered i get:

*CLI>     -- Executing Answer("H323/ip$10.0.3.23:32782/6502", "") in new
stack
    -- Executing Playback("H323/ip$10.0.3.23:32782/6502",
"demo-instruct") in new stack
    -- Playing 'demo-instruct' (language 'en')

which is the normal procedure.
The connexion is well built between the client and asterisk (H225 &
H245) and well negociated with the codec (gsm).

But no RTP stream comes out of the asterisk (I tcpdumped to be sure).

My question is:

1/Is there a way to explain this ? (lack of configuration, compilation
  options)

if not,

2/ Is there a way to investigate deeper in order to understand where
   does the RTP stream faint inside Asterisk ?

regards,


-- 
Kiel