| Summary: | ASTERISK-13619: Rtp socket are not closed after Hangup | ||
| Reporter: | triccyx (triccyx) | Labels: | |
| Date Opened: | 2009-02-20 09:59:56.000-0600 | Date Closed: | 2011-06-07 14:02:36 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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) ast_rtp_destroy(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. | ||