I am using the guide at https://openmeetings.apache.org/AsteriskIntegration.html to implement Asterisk and VOIP.
Before under previous additions, when I entered the room, the SIP transport agent would also enter the room. Now after upgrading from 5.0 to 6.10 when I enter the room no sip transport agent enters. What information do I need to provide to anyone so I can troubleshoot this matter. Is there an upgraded version of this guide https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration ? The sipusers table in 6.1 looks nothing like the table in this guide. Sincerely Bro Miles YAH's Global Kingdom Ministries.
