On Thu, 18 Feb 2010, Karsten Wemheuer wrote: > Am Donnerstag, den 18.02.2010, 10:49 +0100 schrieb Armin Schindler: >> 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. > > With version 1.4.29 I can't reproduce it the way You described it. If > the caller hangs up before * times out the INVITE, the ressources are > freed (SIP-channel and RTP-Ports). If * times out first, the ressources > are freed some time later (< 1 minute).
Yes, I can confirm that. I now have updated the production system to 1.4.29 and the issue seems to be solved. I cannot reproduce it anymore. 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