How to disable / disallow reinvite after SIP OK? I have problem with oneway audio so I need to send RTP via Asterisk
have you try directmedia=no
on my extension number or sip.conf as global ?
There is no extension number on sip.conf , you have sip peers, I guess directmedia it is a global setting , but just in case try both
But you understand i need to send RTP via Asterisk ? I will try
EDIT: Yes, directmedia=no will disable reinvite after answer call. SOLVED
diectmedia can go in both general and specific sections.