Hi Sanjay,

I'm not sure about that, but I think you can configure it in, for
example, /etc/asterisk/sip.conf.
There is an option that you configure for each channel like:

only=gsm

It instructs the sip protocol, that "only" gsm codec must be used.

I hope it has helped you.

Regards,

Ronaldo.

On 4/3/07, Sanjay Rajdev <[EMAIL PROTECTED]> wrote:
Hello All,

I would like to only use the gsm codec for all the calls, is it possible I want 
to use minimum possible bandwidth as we have most of calls over Internet.

Regards,
Sanjay Rajdev

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to