| Summary: | ASTERISK-27847: Asterisk Crashes (Excessive refcount) | ||
| Reporter: | Jestin Philip (jphilip777) | Labels: | pjsip |
| Date Opened: | 2018-05-10 09:50:13 | Date Closed: | 2020-01-14 11:13:52.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/CodecHandling Sounds |
| Versions: | 13.21.0 | Frequency of Occurrence | Frequent |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | We have an asterisk system with 1009 sip extensions. Around 600 are registered.
We get around 100 active channels/50 active calls at the same time. Recently asterisk started to crash frequently with the following error on the CLI [May 10 14:56:42] ERROR[14464][C-0000080f]: astobj2.c:518 __ao2_ref: Excessive refcount 100000 reached on ao2 object 0x12f5628 [May 10 14:56:42] ERROR[14464][C-0000080f]: astobj2.c:518 __ao2_ref: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x12f5628 (0) Got 14 backtrace records #0: [0x60d01e] main/utils.c:2487 __ast_assert_failed() (0x60cf96+88) #1: [0x45d823] main/astobj2.c:437 internal_ao2_ref() #2: [0x45db98] main/astobj2.c:519 __ao2_ref() (0x45db67+31) #3: [0x528113] main/frame.c:348 ast_frdup() (0x527ea7+26C) #4: [0x4641bd] main/audiohook.c:341 audiohook_read_frame_both() #5: [0x464470] main/audiohook.c:396 audiohook_read_frame_helper() #6: [0x46469b] main/audiohook.c:452 ast_audiohook_read_frame_all() (0x464651+4A) #7: [0x7fa936b14ddb] apps/app_mixmonitor.c:693 mixmonitor_thread() #8: [0x609a8f] main/utils.c:1239 dummy_start() We also get a similar error whenever we do a module reload [May 10 14:58:49] ERROR[16049]: astobj2.c:518 __ao2_ref: Excessive refcount 100000 reached on ao2 object 0xc808b8 [May 10 14:58:49] ERROR[16049]: astobj2.c:518 __ao2_ref: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0xc808b8 (0) Got 28 backtrace records #0: [0x60d01e] main/utils.c:2487 __ast_assert_failed() (0x60cf96+88) #1: [0x45d823] main/astobj2.c:437 internal_ao2_ref() #2: [0x45db98] main/astobj2.c:519 __ao2_ref() (0x45db67+31) #3: [0x525541] main/format_cap.c:226 __ast_format_cap_append() (0x5254c9+78) #4: [0x567e95] main/media_index.c:335 update_file_format_info() #5: [0x567f63] main/media_index.c:363 process_media_file() #6: [0x5689bb] main/media_index.c:507 process_file() #7: [0x568da3] main/media_index.c:577 media_index_update() #8: [0x568d52] main/media_index.c:564 media_index_update() #9: [0x568ce7] main/media_index.c:560 media_index_update() #10: [0x568e6d] main/media_index.c:591 ast_media_index_update() (0x568e45+28) #11: [0x5d42cd] main/sounds.c:106 update_index_cb() #12: [0x45e9d9] main/astobj2_container.c:354 internal_ao2_traverse() #13: [0x45ed0b] main/astobj2_container.c:456 __ao2_callback() (0x45ecac+5F) #14: [0x5d43f5] main/sounds.c:138 ast_sounds_reindex() (0x5d42fc+F9) #15: [0x53cba8] main/loader.c:1017 ast_module_reload() (0x53c96e+23A) #16: [0x4d36b0] main/cli.c:274 handle_reload() #17: [0x4dc6cb] main/cli.c:2808 ast_cli_command_full() (0x4dc457+274) #18: [0x7fa960fde53f] res/res_clialiases.c:149 cli_alias_passthrough() #19: [0x4dc6cb] main/cli.c:2808 ast_cli_command_full() (0x4dc457+274) #20: [0x4dc82f] main/cli.c:2835 ast_cli_command_multiple_full() (0x4dc797+98) #21: [0x455631] main/asterisk.c:1549 netconsole() #22: [0x609a8f] main/utils.c:1239 dummy_start() Could it be we just have too many sip extensions ? We use chan_sip and not chan_pjsip. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-05-10 09:50:15.617-0500 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]. By: Joshua C. Colp (jcolp) 2018-05-15 02:45:31.880-0500 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information In your case we need to know more about how you are using Asterisk, what you are doing. A configuration that exhibits the problem would also be useful. By: Asterisk Team (asteriskteam) 2018-05-29 12:00:00.855-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||