El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió: > Hi, I've read that FS supports/implements Session Timers to monitorice > both legs of a call. How to enable it? I mean: > > alice ------- FS -------- bob > > - alice calls bob vía FS > - FS calls bob. > - bob answers (sends 200 OK). > - "bypass_media" mode, no RTP through FS. > - FS establishes a SIP dialog with alice and other one with bob. > - From this moment FS starts sending periodically in-dialog > INVITE/UPDATE to both legs in order to check if each SIP dialog is > still alive in both endpoints. > - In case alice crashes (looses dialog info), alice will reply "481 > Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE > arrives from FS, so FS will understand that alice has ended the dialog > (or has crashed) and sends a BYE to bob. > > Is it possible with FS? how to enable it?
I've found those options in Sofia profiles: <param name="enable-timer" value="false"/> <param name="minimum-session-expires" value="120"/> They seem to be related to SIP Session Timers (nothing related to RTP), am I right? Thanks. -- Iñaki Baz Castillo _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org