Hello, When I do a SIP URI call from my softphone, the call is made directly to the destination host (p2p), bypassing the PBX. So I lose the possibility of recording, making statistics, etc ...
Is there a way to force URI calls through the PBX? I have found no configuration at the client or at the server level. Do you know any softphone that will allows me to do this ? Thank you and have a nice day, Julien -- Julien Sansonnens -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
