| Summary: | ASTERISK-13649: Rtp socket are not closed after Hangup | ||
| Reporter: | triccyx (triccyx) | Labels: | |
| Date Opened: | 2009-02-25 04:36:58.000-0600 | Date Closed: | 2011-06-07 14:03:16 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) full ( 1) portLog ( 2) rpt.conf ( 3) sip.conf | |
| Description: | After a sip call the rtp ports are never closed. ****** ADDITIONAL INFORMATION ****** 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 Missing 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. | ||