How dose Asterisk translate sip-h323?

who can tell me?thks

This is done using channel bridging.
In channel bridging two channels with different technologies are bridged and transcoding is done for the media streams. So in this scenario all the voice traffic should pass through asterisk server that performs the switching.

I understand what you are tlling, but how can I make it to my Asterisk? I have ine mashine with asterisk and sip. To take VoIP with H323 I must have another server with asterisk which work with H323 and to make bridge between them. This is what I’ve understand from your lines. But how can I make it? Thanks

You can do the translation all on one box. You will need to setup a H.323 connection (trunk/extension) and a SIP connection (trunk/extension) to a device, and then use your dial plan (in extensions.conf) to connect the two. Yout don’t have to tell Asterisk to do the conversion.

I have tested such a configuration a while back to connect SIP extensions into a CISCO Call Manager using OpenH323 and gnugk (because I needed an H.323 gatekeeper).

coud you advise your workable config (extensions.conf, sip.conf, h323.conf)?

i have been trying to do sip-to-h323 conversion but without success. The symtom is that no voice heard on both side or either side.

codec is g729r8

pls advise.

Ensure that you have real bind IP address in /etc/asterisk/h323.conf, otherwise it can produce very interesting effects. :wink: