[asterisk-users] SIP RTP ports not released when channel is hung up

[Karsten Wemheuer]

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