[asterisk-users] stucked calls in asterisk 1.4

[Stefan Schmidt]

David Backeberg schrieb:
> On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt <sst at sil.at> wrote:
>> Server A call it PBX there are the sip clients connected
>> A call comes from server B or C to server A and then to a client, gets
>> stucked on Server A when PSTN side hangs up. On server B or C the call
> 
> You may not have properly configured your card for the way to detect
> hangups. If this is going to a proprietary PBX, you may need to change
> around the line signaling. Some kinds of line signaling reverse the
> polarity on the line voltage to signal a hangup. Other lines don't.
> The cheap trick is to try your settings both ways and when one way
> works to use that.

its not a proprietary pbx its just a self developed asterisk and the
server where the card is recognize the hangup, but the bye from the
server (b or c) to the pbx dont work.

as i said the routing server also handles calls from an ser proxy and
another asterisk server where iax accounts terminates and this problem
is only on the pbx server.

Maybe it is a network problem but the quality of the rtp streams is ok
but i think that there are too much sip pakets for the system.

there are around 1600 sip users registerd, 300 - 400 sip channels
(register, options, notifys and invite) and 600 - 700 subscriptions so
there is much sip traffic.

this server also does rtp handling and have 50 to 100 calls (active
ones) and in peek time there is around 10mbit of traffic with 5 to 6 kpps.

maybe a problem with udp buffer size??

thanks for your help so far david!

best regards

steve