Rtcp halted Operation not permitted

[james (James Zhu)]

please check with the port and nat:

Ports being used are out of range set in rtp.conf

For anyone else that gets similar issues to what I had, I wanted to let you know how I fixed it. In freepbx, go to Tools and Asterisk SIP Settings. In Media and RTP Serttings, I changed Reinvite Behavior to “Update.” The RTCP errors appear to have been unrelated to the initial issue of one way audio (happening only when greeting played before transfer to extension).