Summary:ASTERISK-13619: Rtp socket are not closed after Hangup
Reporter:triccyx (triccyx)Labels:
Date Opened:2009-02-20 09:59:56.000-0600Date Closed:2011-06-07 14:02:36
Versions:Frequency of
Description:After a sip call the rtp ports are never closed.
Please try with "netstat -lnp | grep asterisk"

Perhaps dialog_unlink_all() need
if (dialog->rtp)

Comments:By: triccyx (triccyx) 2009-02-23 05:12:57.000-0600

Problems also with dialog->stimer "too many files open" and memory leack.

By: Joshua C. Colp (jcolp) 2009-02-23 10:34:02.000-0600

I'm afraid I'm going to need much more information to try to track this down. I've tried many scenarios with both 1.6.1-rc1 and 1.6.1 SVN and they all work fine. Can you provide console output, dialplan, SIP traces, SIP history, everything that could possibly help?

By: triccyx (triccyx) 2009-02-23 10:41:38.000-0600

I use only autocreated peers.
Tomorrow I will send you others output.
I think as you told me it is related to 0014522.

By: Joshua C. Colp (jcolp) 2009-02-23 10:43:07.000-0600

If you truly are using 1.6.1-rc1 then that is not true, as the issue in 14522 seems to be isolated to changes only made in trunk. If you are not using 1.6.1-rc1 then it is important to know exactly what you are indeed using.

By: triccyx (triccyx) 2009-02-23 14:55:52.000-0600

Sorry for mistake. I'm currently on trunk.

By: Joshua C. Colp (jcolp) 2009-02-23 15:37:58.000-0600

Closed per other issue. Fixed in SVN.