-
Notifications
You must be signed in to change notification settings - Fork 449
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
Segfault because of mem_deref(call) in module menu ua_event_handler #1059
Comments
cspiel1
added a commit
to cspiel1/baresip
that referenced
this issue
Aug 13, 2020
A segfault occurs in modules loaded after menu and have an ua_event_handler that use call object (to collect information about the call). The call object may not be de-referenced in any ua_event_handler.
cspiel1
added a commit
to cspiel1/baresip
that referenced
this issue
Aug 15, 2020
A segfault occurs in modules loaded after menu and have an ua_event_handler that use call object (to collect information about the call). The call object may not be de-referenced in any ua_event_handler.
alfredh
pushed a commit
that referenced
this issue
Aug 15, 2020
fixed in commit f420218 |
wip-sync
pushed a commit
to NetBSD/pkgsrc-wip
that referenced
this issue
Nov 28, 2020
(Presumably the changelog [Unreleased] section is for 1.0.0) = Baresip Changelog == [Unreleased] === Added - aac: add AAC_STREAMTYPE_AUDIO enum value - aac: add AAC_ prefix - Video mode param to call_answer(), ua_answer() and ua_hold_answer [#966] - video_stop_display() API function [#977] - module: add path to module_load() function - conf: add conf_configure_buf - test: add usage of g711.so module [#978] - JSON initial codec state command and response [#973] - account_set_video_codecs() API function [#981] - net: add fallback dns nameserver [#996] - gtk: show call_peername in notify title [#1006] - call: Added call_state() API function that returns enum state of the call [#1013] - account_set_stun_user() and account_set_stun_pass() API functions [#1015] - API functions account_stun_uri and account_set_stun_uri. [#1018] - ausine: Audio sine wave input module [#1021] - gtk/menu: replace spaces from uri [#1007] - jack: allowing jack client name to be specified in the config file [#1025] [#1020] - snapshot: Add snapshot_send and snapshot_recv commands [#1029] - webrtc_aec: 'extended_filter' config option [#1030] - avfilter: FFmpeg filter graphs integration [#1038] - reg: view proxy expiry value in reg_status [#1068] - account: add parameter rwait for re-register interval [#1069] - call, stream, menu: add cmd to set the direction of video stream [#1073] === Changed - **Using [baresip/re](https://github.com/baresip/re) fork now** - audio: move calculation to audio_jb_current_value - avformat: clean up docs - gzrtp: update docs - account: increased size of audio codec list to 16 - video: make video_sdp_attr_decode public - config: Derive default audio driver from default audio device [#1009] - jack: modifying info message on jack client creation [#1019] - call: when video stream is disabled, stop also video display [#1023] - dtls_srtp: use tls_set_selfsigned_rsa with keysize 2048 [#1062] [#1056] - rst: use a min ptime of 20ms - aac: change ptime to 4ms === Fixed - avcodec: fix H.264 interop with Firefox - winwave: waveInGetPosition is no longer supported for use as of Windows Vista [#960] - avcodec: call av_hwdevice_ctx_create before if-statement - account: use single quote instead of backtick - ice: fix segfault in connh [#980] - call: Update call->got_offer when re-INVITE or answer to re-INVITE is received [#986] - mk: Test also for /usr/lib64/libspeexdsp.so to cover Fedora/RHEL/CentOS [#992] - config: Allow distribution specific CA trust bundle locations (fixes [#993] - config: Allow distribution specific default audio device (fixes [#994] - mqtt: fix err is never read (found by clang static analyzer) - avcodec: fix err is never read (found by clang static analyzer) - gtk: notification buttons do not work on Systems [#1012] - gtk: fix dtmf_tone and add tones as feedback [#1010] - pulse: drain pulse buffers before freeing [#1016] - jack: jack_play connect all physical ports [#1028] - Makefile: do not try to install modules if build is static [#1031] - gzrtp: media_alloc function is missing [#1034] [#1022] - call: when updating video, check if video stream has been disabled [#1037] - amr: fix length check, fixes [#1011] - modules: fix search path for avdevice.h [#1043] - gtk: declare variables C89 style - config: init newly added member - menu: fix segfault in ua_event_handler [#1059] [#1061] - debug_cmd: fix OpenSSL no-deprecated [#1065] - aac: handle missing bitrate parameter in SDP format - av1: properly configure encoder === Removed - ice: remove support for ICE-lite - ice: remove ice_debug, use log level DEBUG instead - ice: make stun server optional - config: remove ice_debug option (unused) === Contributors (many thanks) - Alfred E. Heggestad - Alexander Gramner - Andrew Webster - Christian Spielberger - Christoph Huber - Davide Alberani - Ethan Funk - Juha Heinanen - mbattista - Michael Malone - Mikl Kurkov - ndilieto - Robert Scheck - Roger Sandholm - Sebastian Reimers [#966]: baresip/baresip#966 [#977]: baresip/baresip#977 [#978]: baresip/baresip#978 [#973]: baresip/baresip#973 [#981]: baresip/baresip#981 [#996]: baresip/baresip#996 [#1006]: baresip/baresip#1006 [#1013]: baresip/baresip#1013 [#1015]: baresip/baresip#1015 [#1018]: baresip/baresip#1018 [#1021]: baresip/baresip#1021 [#1007]: baresip/baresip#1007 [#1025]: baresip/baresip#1025 [#1020]: baresip/baresip#1020 [#1029]: baresip/baresip#1029 [#1030]: baresip/baresip#1030 [#1038]: baresip/baresip#1038 [#1009]: baresip/baresip#1009 [#1019]: baresip/baresip#1019 [#1023]: baresip/baresip#1023 [#1062]: baresip/baresip#1062 [#1056]: baresip/baresip#1056 [#960]: baresip/baresip#960 [#980]: baresip/baresip#980 [#986]: baresip/baresip#986 [#992]: baresip/baresip#992 [#993]: baresip/baresip#993 [#994]: baresip/baresip#994 [#1012]: baresip/baresip#1012 [#1010]: baresip/baresip#1010 [#1016]: baresip/baresip#1016 [#1028]: baresip/baresip#1028 [#1031]: baresip/baresip#1031 [#1034]: baresip/baresip#1034 [#1022]: baresip/baresip#1022 [#1037]: baresip/baresip#1037 [#1011]: baresip/baresip#1011 [#1043]: baresip/baresip#1043 [#1059]: baresip/baresip#1059 [#1061]: baresip/baresip#1061 [#1065]: baresip/baresip#1065 [#1068]: baresip/baresip#1068 [#1069]: baresip/baresip#1069 [#1073]: baresip/baresip#1073 [Unreleased]: baresip/baresip@v0.6.6...HEAD
Sign up for free
to join this conversation on GitHub.
Already have an account?
Sign in to comment
In case of UA_EVENT_CALL_TRANSFER_FAILED, UA_EVENT_AUDIO_ERROR the pointer call given as function parameter is de-referenced.
Other ua_event_handler that use information from call will crash with a segfault if there are called after menus ua_event_handler.
It depends on the order of modules initialization. If a module with an ua_event_handler is loaded after menu, the segfault occurs.
Commits that cause the issue:
commit 7b6b8a0
Author: Alfred E. Heggestad alfred.heggestad@gmail.com
Date: Wed Oct 10 17:57:26 2018 +0200
commit 4dd9bc9
Author: juha-h jh@tutpro.com
Date: Tue Oct 30 11:42:29 2018 +0200
The text was updated successfully, but these errors were encountered: