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Summary:ASTERISK-28286: chan_sip - no lock pvt data in proc_session_timer()
Reporter:Krzysztof Trempala (krzysztoft)Labels:
Date Opened:2019-02-13 07:05:29.000-0600Date Closed:
Priority:MajorRegression?
Status:Open/NewComponents:Channels/chan_sip/General
Versions:13.24.1 Frequency of
Occurrence		Frequent
Related
Issues:
Environment:Ubuntu 12.04Attachments:
Description:I have a problem with accidentally overwriting the heap in the situation:
- process session refresh timeout in function proc_session_timer() run transmit_reinvite_with_sdp()
- call has been terminated and channel is destroying

{noformat}
+====================================+
|         sched thread               |
+====================================+
..
proc_session_timer()
 transmit_reinvite_with_sdp (p, ..)
  try_suggested_sip_codec(p,..)              +=========================+
   pbx_builtin_getvar_helper(chan=p->owner)  |   channel thread        |
                                             +=========================+
                                             | ...                    |
           <---------------------------------| sip_hangup              |
                                             |  sip_set_owner(p, NULL);|
                                             |   p->owner = NULL;      |
                                             | ...                     |
                                             | channel destroy         |
    ast_channel_lock(chan);                  +-------------------------+

{noformat}
Comments:By: Asterisk Team (asteriskteam) 2019-02-13 07:05:30.972-0600

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].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.
By: Chris Savinovich (csavinovich) 2019-02-18 11:20:55.005-0600

Hello Krzysztof,
Starting with versions 13.x, Asterisk is in the process of solely supporting chan_pjsip for SIP calls and allowing chan_sip to be solely supported  by the community. You would be more than welcome to submit any patches to chan_sip using our Gerrit site that you estimate can fix this issue.

Thanks