What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec.
Thanks, Scott On 9/14/07, Scott Moseman <[EMAIL PROTECTED]> wrote: > > I have a fresh 1.4.10.1 installation that appears to have a problem > with g729 pass-through. I can see the gateway in question sending > an INVITE using g729b. However, the Asterisk is only sending g711 > in the INVITE to my Polycom phone. > > [gateway] > disallow=all > allow=g729 > > [phone] > disallow=all > allow=ulaw > allow=alaw > allow=g729 > > There's the codec configs for the gateway and the phone in question. > I even attempted to setup the phone with only the allow=g729, but in > that instance it won't even complete the call. We had to add g711 > support to the gateway in question for now to get it up and running, > but we want to get it back to using only g729. > > CLI> show modules like g729 > Module Description > Use Count > format_g729.so Raw G729 data > 0 > codec_g729a.so Annex A/B (floating point) G.729 Codec > ( 0 > 2 modules loaded > > I downloaded the pre-compiled g729 module from Digium. The sip.conf > references g729 and the codec module is loaded. Unless there's > anything else I need to do that I'm missing? > > Thanks, > Scott > _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
