[asterisk-users] stucked calls in asterisk 1.4

[Stefan Schmidt]

David Backeberg schrieb:
> Now that I better understand your problem, I'm out of ideas.

thats the point where i stand ;)

> You are correct that if a BYE sip packet gets lost,
> a) it won't get retransmitted if it's UDP
> b) the side that's waiting for the hangup will think the call is still active
> 
> I've seen this in my system where the network switch went down while
> calls were active. The system where the call was happening caught the
> hangup, but the trunk system never got the bye as the network was
> down.
> 
> My only questions are:
> Are you using a quality network switch, or maybe you can use a
> cross-over cable to eliminate collisions with other systems?
> 
all server are in one rack in our datacenter and are connected to an HP
Procurve 2650 switch, which has been setup around 3 months ago, cause of
the old switch died silent in the night.

all server had two interfaces and i have allready tried to route the
traffic between the pbx and the routing server over the second
interface, where database requests normally run. But this didnt solved
the problem too.

i will try to increase the UDP buffer size in the linux kernel, maybe
this will take some affect.

best regards

steve