[asterisk-users] remove

[Julio lopez]

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users at rogg.is
Sent: Saturday, June 09, 2007 5:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to tell what codec is used for each end of a
call MD110->H323->SIP

 

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?

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

 

Thank you for your time and effort to respond.

 

Baldvin

 

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