(that cc'ing is just too hard for me, it seems! ;-)).
It's already been located to a specific customer made UA, so we are trying to find out what goes wrong there. It seems to be a problem with the UA (softphone).
Thanks.
On Tue, Oct 27, 2009 at 4:42 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 27.10.2009 21:22 Uhr, Anders wrote:
That was exactly the problem Daniel - no BYE was ever sent from the UA, so that's what we need to fix!
please keep cc-ing the mailing list.
You can fix the problem by identifying the sip devices that do not send BYE. You are doing record routing, right?
In 1.5, dialog can send BYE at call timeout -- does not help much you, but can close eventual open channels in gateways.
The missing BYE happens when call is between two SIP phones? Or between sip phone and pstn gateway/media server?
Cheers, Daniel
Thanks!!
On Tue, Oct 27, 2009 at 4:16 PM, Daniel-Constantin Mierla miconda@gmail.com wrote:
On 26.10.2009 16:41 Uhr, Anders wrote:
Hi,
I have two issues, and I think they are connected. The number of Active Dialogs keeps growing - as if some of them are hung. Not all of them, but some of them. At the same time, I have seen that from a specific customer, there is no BYE in the 'acc' table in the accounting. So, my conclusion - no BYE means it's not finished means it's hung... - right?
Any ideas where to look?
if you don't get a BYE in acc table then maybe was not sent and that keeps the dialog active. You can set a max time per call -- timeout -- for each dialog:
http://kamailio.org/docs/modules/1.5.x/dialog.html
However, is good to identify why BYE is not coming, maybe is a fraud attempt or a broken sip device.
Cheers, Daniel
-- Daniel-Constantin Mierla
- Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
- http://www.asipto.com/index.php/sip-router-masterclass/
-- Daniel-Constantin Mierla
- Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
- http://www.asipto.com/index.php/sip-router-masterclass/