On Wednesday 09 August 2006 01:26, Chuck Mariotti wrote: > Do you have any idea of how to test if passing the call to a remote SIP > Softphone client is working? Is there a test or something I can do to > verify this is happening now? Or does it just "happen" automatically?
I'm not sure I understand the question. Do you want to test to see if the RTP is passing through your system? I am not aware of anything in the Asterisk CLI that tells you this, but running ethereal on that network device should tell you. :-) -A.
