asterisk users - Apr 2005 - sip <-> oh323 / real-time / g729 - one way audio

Hi,

I am using real-time, oh-0.7.2, G729

Calling from (SIP)UA through asterisk towards h323 devices or the other way
round, I get only one-way audio.

Called party can only talk, caller can only listen. 

Calling SIP to SIP is ok.

All devices are on official IP addresses.
(no NAT)


Regards,  

Shaoul Jacobson
Senior VoIP Consultant
Tellink
Tel :	+32 3 201 96 36
Fax : 	+32 3 227 09 81
e-mail	shaoul@tellink.com

Are the h323 devices on public IP or behind NAT?


Selon Shaoul Jacobson - TELLINK <shaoul@tellink.com>:
>
>
> Hi,
>
> I am using real-time, oh-0.7.2, G729
>
> Calling from (SIP)UA through asterisk towards h323 devices or the other way
> round, I get only one-way audio.
>
> Called party can only talk, caller can only listen.
>
> Calling SIP to SIP is ok.
>
> All devices are on official IP addresses.
> (no NAT)
>
>
> Regards,
>
> Shaoul Jacobson
> Senior VoIP Consultant
> Tellink
> Tel :	+32 3 201 96 36
> Fax : 	+32 3 227 09 81
> e-mail	shaoul@tellink.com
>
> _______________________________________________
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>

Is there a good place to get the command references for the Extensions.conf? 
The Wiki and other documentation seem to be rather limited and don't explain
all the parameters.


Regards,

Chris