Hi, did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It looks related to your issue.
Best regards, Marcus On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler <[email protected]> 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 - [email protected] 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
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