Hi guys,If someone could help me, with asterisk sip installation, I've setup the sip regarding this documentation http://openmeetings.apache.org/red5sip-integration_3.0.html But i dont know if it's working or not !!! how we can check ? How we link current user to their sip phone ? Is it normal, there is not field for sip in user configuration ? i'm using version 3.0.5 snapshot
regards
