[EMAIL PROTECTED] 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.
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