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
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
What version of asterisk-oh323 do you use? Michael. 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 ? > > regards, > >
On Fri, Jun 25, 2004, Michael Manousos wrote:> > What version of asterisk-oh323 do you use?As I said the one checkouted recently (Jun 23 2004 or something like this) -- Kiel
I am in the process of designing a box to provide VoIP integration into the PSTN via our Soft Switch. It seems to me that going from a PRI to VoIP would involve floating point operations, making hyperthreading of benefit. Is this correct ? -- James H. Edwards Routing and Security Administrator At the Santa Fe Office: Internet at Cyber Mesa jamesh@cybermesa.com noc@cybermesa.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040627/e762120d/attachment.pgp