Summary: | ASTERISK-25226: chan_sip: Channel leak in branch 13 on early replaces call pickup | ||
Reporter: | Walter Doekes (wdoekes) | Labels: | |
Date Opened: | 2015-07-02 09:08:08 | Date Closed: | 2015-07-04 19:12:23 |
Priority: | Major | Regression? | Yes |
Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
Versions: | 13.4.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | When doing a (non-magic) early call pickup, the Asterisk 13 branch chan_sip leaks channels.
Taking the callpickup.sh from ASTERISK-25213, you observe a lost channel after a single run: {noformat} *CLI> core show channels Channel Location State Application(Data) SIP/bob-00000001 (None) Ringing AppDial((Outgoing Line)) 0 active channels 0 active calls 1 call processed {noformat} Valgrind reports the leaks as coming from sip_new(). Also note that {{core stop gracefully}} refuses to stop because of the leaked channel. The patch at gerrit fixes it. Thanks Matt and Corey for looking at the other (bridge) path portion of the patch. Cheers, Walter Doekes OSSO B.V. | ||
Comments: | By: Asterisk Team (asteriskteam) 2015-07-02 09:08:10.736-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. |