Hi, 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). Karsten -- _____________________________________________________________________ -- 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