Hi,<br><br>did you see this one: <a href="https://issues.asterisk.org/view.php?id=16774">https://issues.asterisk.org/view.php?id=16774</a> ? It looks related to your issue.<br><br>Best regards, Marcus<br><br><div class="gmail_quote">
On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler <span dir="ltr"><<a href="mailto:armin@melware.de">armin@melware.de</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im">On Fri, 12 Feb 2010, Armin Schindler wrote:<br>
>>>> I had a look at<br>
>>>> netstat -nuap<br>
>>>> and it shows that a lot of ports are still assigned, even if there is no<br>
>>>> channel in use.<br>
>>>> But "sip show channels" show a lot of (unused) entries with no<br>
>>>> codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.<br>
>> REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If<br>
>> you have a SIP channel that has a last message being INVITE and still say<br>
>> you have no calls, you have a problem right there.<br>
><br>
> I just see these entries with "sip show channels", but cannot tell if<br>
> e.g. the REGISTER listed channels have RTP ports allocated.<br>
> Who can I find out which SIP channel allocated which port?<br>
> Or which SIP channel belongs to the ports I see with 'netstat -nuap'?<br>
<br>
</div>I just made a test to confirm:<br>
After a restart of asterisk (to have a clean state with no sip channels<br>
activ and no RTP port allocated), I can confirm that:<br>
- REGISTER and OPTION listed sip channels don't use RTP ports<br>
- after some calls (e.g. SIP to SIP) the RTP ports are freed immediately<br>
(looks like this is the case on hangup before answer).<br>
- after some other calls, the RTP ports are freed after about 20-30 seconds<br>
after hangup.<br>
So basically all is correct.<br>
<div class="im"><br>
> I do have a sip channels like<br>
> 172.21.4.114 666 0430c3a638e 00102/00000 0x0 (nothing) No Init: INVITE<br>
> in 'sip show channels' and they don't go away for a long time.<br>
> Shouldn't there be a timeout to destroy such a channel even if somehow<br>
> the phone was 'disconnected' in during a call?<br>
><br>
>>> If the channels exists even after 32 seconds after BYE, and BYE was<br>
>>> signaled correctly, I would file a bug report.<br>
<br>
</div>It really looks like that there is a case where the sip channel is not<br>
destroyed and that is the cause of the problem.<br>
I will try to reproduce this.<br>
Any ideas?<br>
<div><div></div><div class="h5"><br>
Armin<br>
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Dipl.-Inf. (FH)<br>Marcus Hunger - <a href="mailto:hunger@sipgate.de">hunger@sipgate.de</a><br>Telefon: +49 (0)211-63 55 55-61<br>Telefax: +49 (0)211-63 55 55-22<br>
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