Hi
I am trying to send call from webrtc to freeswitch and then bridge to an external endpoint that is a normal SIP phone. I tried to use sip_secure_media=false to force the media to be converted from SAVP to AVP.
But the outgoing INVITE is still showing SAVP ( sip trace blow ). Does anyone know where I am doing wrong? any help will be greately appreciated.
2015-08-17 15:29:46.341244 [INFO] mod_dialplan_xml.c:637 Processing 1001 <1001>->1002 in context default