[asterisk-users] How to tell what codec is used for each end of a call MD110->H323->SIP

[mail-lists]

asterisk-users at rogg.is wrote:
> Hi.
> 
>  
> 
> Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the
> call established but no sound heard on either end.
> 
>  
> 
> What is the best/correct way to try and see what codecs Asterisk is using on
> each end of the call as it passes through Asterisk

for SIP I use 'sip show channels' I'm not sure what the equivilent h323 
command is.


> 
> And is there any way to see that voice is in fact being passed through
> Asterisk during the call (some counters etc.)?
> 

Try 'rtp debug' and the rtp packets should scroll by.
>  
> 
> Thank you for your time and effort to respond.
>