hello, i have a problem with stucked or hanging calls in asterisk 1.4.25 we had this problem before and so we upgradet from 1.2.32 to 1.4.25 but it still exists and as i could see, happens even more.
on this server there are > 1500 clients registered all with qualify on and we had 2 routing server with 4 E1 pstn connects on each. The connect between this server and the 2 routing server runs over sip. This problem only appears on this server and not on the routing server. Even if i soft hangup a stucked channel i get the sip response 481 call/leg transaction doesnt exists back from the routing server, so one call leg had allready send a bye but this server hasnt closed the call. i think there could be a network problem but also the server itself and the switch where the 3 servers are connected is younger than 3 months and we had the same problem with this system on an older server. sometimes i see that most of the sip peers get unreachable or too lagged so i think that there could be a problem for asterisk to handle that amount of pakets. i´ve just splittet up the traffic on 2 interfaces, so that normal traffic from the clients comes to eth0 and the traffic from and to the routing servers runs over eth1 but the problem still occurs. do you have any ideas what i could do to solve this problem? best regards steve smith _______________________________________________ -- 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