Summary:ASTERISK-13649: Rtp socket are not closed after Hangup
Reporter:triccyx (triccyx)Labels:
Date Opened:2009-02-25 04:36:58.000-0600Date Closed:2011-06-07 14:03:16
Versions:Frequency of
Environment:Attachments:( 0) full
( 1) portLog
( 2) rpt.conf
( 3) sip.conf
Description:After a sip call the rtp ports are never closed.


This is the repetition of id=0014519
I tried the new version from trunk for id=0014522 but it still doesn't work.
Comments:By: triccyx (triccyx) 2009-02-25 04:46:02.000-0600

I only use autocreated peers.
In the full log there are some log of mine you can ignore them.
portlog  cames from "netstat -lnp | grep asterisk > /usr/src/portLog"

The test has been done with 18 simpleopan phones that do loop calls

By: Joshua C. Colp (jcolp) 2009-02-25 08:46:09.000-0600

I just spent some time trying to reproduce this without the configs provided and with the configs provided. Both worked fine and the dialog was destroyed and RTP sockets closed. After looking at your full log I see that you have made some considerable changes to chan_sip. I'm going to have to ask you to try things on a vanilla installation of Asterisk with no modifications.

By: triccyx (triccyx) 2009-02-25 10:31:04.000-0600

dialog_unref(cur, "drop ref in iterator loop break");
in my code sorry again

By: Joshua C. Colp (jcolp) 2009-02-25 10:32:43.000-0600

Closed per reporter, issue in outside modified code.