[Asterisk-Users] Failure in RTP streaming

[kiel hedjam]

On Fri, Jun 25, 2004, kiel hedjam wrote:
> 
> 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 ?


The version I used was the last cvs snapshot, I've just been trying with
the 0.9.0 (the tar.gz version) and evrything is all right.

I don't why I didn't get RTP streams with the cvs version, if I got time
I would investigate a little bit. If anybody here have an idea ...

-- 
Kiel