Right now when my Asterisk transfers an inbound sip call to an external number, also sip, it keeps the first call and starts a new one.
I would like to make a release of that call so that the RTP doesn’t go through my Asterisk. Is that possible? And if so, how?
I’m running Asterisk on Debian and my Asterisk is 1.2. I’m also running Freepbx right now. 188.8.131.52.
Thank you for your time.
Kind Regards, Henrik