On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt <[email protected]> wrote: > i have a problem with stucked or hanging calls in asterisk 1.4.25 > only appears on this server and not on the routing server. Even if i
I'm confused. So the server where the calls get stuck has both SIP and DAHDI/Zaptel channel calls? And the SIP side of those calls isn't getting hungup? It is true that if your server doesn't receive a BYE packet it will think that the call is still there. Does your dialplan that attaches to the routing server do a hangup at the end of the call? _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
