Is there a way to originate a call to a specified destination from the asterisk console when we have to test a SIP connected on a remote Asterisk server?
I have this requirement when we have to test a new SIP connection is working on a remote Asterisk server. This is a case where we do not have a way to originate a call from a handset on the local site. I believe we will be able to confirm that it is working when we enable a debug with "sip debug peer " on the Asterisk console.
Thank you for the reply