asterisk users - May 2009 - stucked calls in asterisk 1.4

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

On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt <sst at sil.at>
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?