asterisk users - Jun 2007 - 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070609/f6510f5d/attachment.htm

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
asterisk-users@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

 

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.8.13/841 - Release Date: 6/9/2007
8:52 AM


No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.472 / Virus Database: 269.8.13/841 - Release Date: 6/9/2007
8:52 AM
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20070609/0604d368/attachment.htm

asterisk-users@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.
>