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

[Armin Schindler]

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