| Summary: | ASTERISK-19779: Asterisk segfaults when handling sip_security_event and attempting to load realtime peer with no realtime backend | ||||
| Reporter: | Aleksandr Gordeev (axonaro) | Labels: | |||
| Date Opened: | 2012-04-24 12:15:56 | Date Closed: | 2015-03-13 21:39:24 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Core/PBX | ||
| Versions: | 10.3.0 10.4.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) ASTERISK-19779-backtrace.txt | |||
| Description: | Asterisk always segfault after processing a certain number of calls extensions.conf [test-sip] exten => service,1,Progress exten => service,n,Wait(2) exten => service,n,Playback(vm-sorry,noanswer) exten => service,n,Hangup | ||||
| Comments: | By: Matt Jordan (mjordan) 2012-04-24 12:20:54.752-0500 Attached backtrace is from the original issue, illustrating the crash originating from the sip_find_peer call from sip_security_event By: Michael L. Young (elguero) 2012-04-24 19:29:04.162-0500 Initial look at this... sip_security_event calls sip_find_peer which calls sip_find_peer_full. This function should find the peer and return it before reaching the code block that proceeds to check for a realtime peer. I started to put together a patch for sip_security_event to check if realtime was being used and then call sip_find_peer with realtime set to TRUE or FALSE but I think something else must be going on. I am not sure why we are reaching that block of code to begin with in sip_find_peer_full. Also, I see other areas of chan_sip that are calling sip_find_peer in the same way as sip_security_event without checking if realtime is being used or not. By: Matt Jordan (mjordan) 2012-04-25 07:29:29.275-0500 Yeah, I'm not convinced this is the fault of sip_security_event, just that the crash originated from there. Based on the backtrace, however, it appears as if an attempt was made to load a non-existant realtime backend from that call - so I'm not sure what got attempted from sip_find_peer that shouldn't have. By: Matt Jordan (mjordan) 2015-03-13 21:39:15.831-0500 Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions | ||||