Summary: | ASTERISK-15242: transmit_refer leaks sip_refer structures | ||
Reporter: | David Woolley (davidw) | Labels: | |
Date Opened: | 2009-11-30 11:40:00.000-0600 | Date Closed: | 2014-11-17 09:58:32.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_sip/Transfers |
Versions: | SVN 11.13.1 12.6.1 13.0.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | transmit_refer is used to implement the sip_transfer function on Up calls. It calls sip_refer_allocate without checking to see if there is already a p->refer structure.
Transfers can fail, so this can happen several times in a call. The only time that this structure is released is when the SIP private structure is destroyed. ****** ADDITIONAL INFORMATION ****** Identified by code reading only and initially on 1.6.1.0, whilst working on ASTERISK-15145, but also present in the current trunk version. This is on my list for ASTERISK-15145, but we may yet abandon it, so I am recording it as as separate issue. (Also the fix may be obvious to someone who knows the code - my slight concern at the moment is that something may be assuming a clean structure. | ||
Comments: | By: Corey Farrell (coreyfarrell) 2014-11-10 00:25:19.792-0600 It looks like this leak is possible, so I've posted a patch to [review board|https://reviewboard.asterisk.org/r/4160/]. |