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
