On Tue, 16 Feb 2010, Armin Schindler wrote:
On Tue, 16 Feb 2010, Marcus Hunger wrote:
Hi,
did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.
Oh thanks, I missed that one.
It really looks related. I have added a note.
Now I know how to reproduce the problem. I added this as note to 16774 as
well:
Start SIP client to register at asterisk, then disconnect that SIP phone
from network. In the time the registration is still active in asterisk, call
this phone. Asterisk will send INVITEs (of course with no answer), then
hangup after about 30 seconds. The asterisk channels are released, but the
sip channel for that "Init: INVITE" is not released.
For now, I can confirm this with 1.4.28 only as I have not tested other
versions yet.
Armin
Best regards, Marcus
On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler <ar...@melware.de> wrote:
On Fri, 12 Feb 2010, Armin Schindler wrote:
>>>> I had a look at
>>>> netstat -nuap
>>>> and it shows that a lot of ports are still assigned, even if
there is no
>>>> channel in use.
>>>> But "sip show channels" show a lot of (unused) entries with no
>>>> codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
>> REGISTER and OPTIONS allocate no RTP ports, so those are not a
problem. If
>> you have a SIP channel that has a last message being INVITE and
still say
>> you have no calls, you have a problem right there.
>
> I just see these entries with "sip show channels", but cannot tell
if
> e.g. the REGISTER listed channels have RTP ports allocated.
> Who can I find out which SIP channel allocated which port?
> Or which SIP channel belongs to the ports I see with 'netstat
-nuap'?
I just made a test to confirm:
After a restart of asterisk (to have a clean state with no sip channels
activ and no RTP port allocated), I can confirm that:
- REGISTER and OPTION listed sip channels don't use RTP ports
- after some calls (e.g. SIP to SIP) the RTP ports are freed immediately
(looks like this is the case on hangup before answer).
- after some other calls, the RTP ports are freed after about 20-30 seconds
after hangup.
So basically all is correct.
> I do have a sip channels like
> 172.21.4.114 666 0430c3a638e 00102/00000 0x0 (nothing) No
Init: INVITE
> in 'sip show channels' and they don't go away for a long time.
> Shouldn't there be a timeout to destroy such a channel even if somehow
> the phone was 'disconnected' in during a call?
>
>>> If the channels exists even after 32 seconds after BYE, and BYE was
>>> signaled correctly, I would file a bug report.
It really looks like that there is a case where the sip channel is not
destroyed and that is the cause of the problem.
I will try to reproduce this.
Any ideas?
Armin
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Dipl.-Inf. (FH)
Marcus Hunger - hun...@sipgate.de
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22
sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391
www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
