On 27.10.2009 21:51 Uhr, Anders wrote:
(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).
self-made UAs can be "tweaked" so they do NOT send BYE so the
accounting
is not closed, hoping that the call won't be billed. It is important in
case you chanrge for some calls, that one side is controlled by you or a
trusted party (e.g., pstn termination provider) and you can set there
some session timeout.
SIP to SIP calls are either free or included in a flat rate, so not very
critical to bill. If you charge them, then is better to use a b2bua,
such as sems, freeswitch or asterisk for such calls and set there a RTP
timeout.
Cheers,
Daniel
Thanks.
On Tue, Oct 27, 2009 at 4:42 PM, Daniel-Constantin Mierla
<miconda(a)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(a)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/
>
>
>
--
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
*
http://www.asipto.com/index.php/sip-router-masterclass/