Summary: | ASTERISK-30489: Asterisk crashed unexpected event I tried to upgrade to latest version | ||
Reporter: | Pham Huu Phu (jonathanpham) | Labels: | webrtc |
Date Opened: | 2023-04-06 22:56:32 | Date Closed: | 2023-05-11 20:51:00 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | Core/General |
Versions: | 20.0.1 | Frequency of Occurrence | Frequent |
Related Issues: | |||
Environment: | Debian created on Nutanix as VMs Debian 10.3 - Asterisk Asterisk 20.2.0 System memory 8GB, 4 x vCPU | Attachments: | ( 0) 2023-04-11_17-43-43.png ( 1) core-asterisk-2023-04-11T01-42-39Z-brief.txt ( 2) core-asterisk-2023-04-11T01-42-39Z-full.txt ( 3) core-asterisk-2023-04-11T01-42-39Z-info.txt ( 4) core-asterisk-2023-04-11T01-42-39Z-locks.txt ( 5) core-asterisk-2023-04-11T01-42-39Z-thread1.txt ( 6) core-asterisk-2023-04-25T10-35-16Z-brief.txt ( 7) core-asterisk-2023-04-25T10-35-16Z-full.txt ( 8) core-asterisk-2023-04-25T10-35-16Z-info.txt ( 9) core-asterisk-2023-04-25T10-35-16Z-locks.txt (10) core-asterisk-2023-04-25T10-35-16Z-thread1.txt (11) debug_log_full.txt (12) extensions.conf.txt (13) full_crashed_25.txt |
Description: | Dear Asterisk team,
We have been using Asterisk for Call Center Call for a long time. We sometimes get problem with Asterisk crashed without any reason. We checked log and find asterisk[46017]: segfault. Our production has CRM connect to Asterisk to make outgoing all. We're stable for a long time If we have about 10 - 15 caller; asterisk services crashed while in calling. We exported backtrack. I hope Asterisk team find a bugs make Asterisk crashed. Thank you very much. | ||
Comments: | By: Asterisk Team (asteriskteam) 2023-04-06 22:56:35.181-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Joshua C. Colp (jcolp) 2023-04-07 03:42:00.254-0500 We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Joshua C. Colp (jcolp) 2023-04-07 03:42:19.406-0500 Additionally, your issue report is confusing. Did this happen after upgrading, or was it already happening and you just tried to upgrade to resolve it? By: Pham Huu Phu (jonathanpham) 2023-04-09 22:01:34.635-0500 Dear Joshua, We got this problem sometime since Asterisk 18.x. We have tried to upgrade to latest version solve this problem. I attached debug file .txt and with extension.conf. Events crashed on Asterisk service crashed at [Apr 6 17:26:46] Have a nice day! By: Joshua C. Colp (jcolp) 2023-04-10 04:14:17.065-0500 This debug log does not include debug. Did you follow the instructions exactly? By: Pham Huu Phu (jonathanpham) 2023-04-10 21:27:22.037-0500 Dear Joshua, Thank you for your help. I updated backtrace and debug log for 2023-04-10 (I limit debug log to +-3 min before/after crashed event). Event crashed time Apr 10 16:53:09 voipmysitekernel: show_signal_msg: 8 callbacks suppressed Apr 10 16:53:09 voipmysitekernel: asterisk[38136]: segfault at 76e267b8 ip 000055ff74bce12e sp 00007f64a4140a40 error 4 in asterisk[55ff74b9e000+1e4000] Apr 10 16:53:09 voipmysitekernel: Code: 7d 98 31 d2 45 85 c0 75 1e e9 6c 02 00 00 66 c7 01 ff 7f 83 c2 01 48 83 c1 02 48 83 c7 02 41 39 d0 0f 84 53 02 00 00 0f bf 37 <44> 0f bf 09 44 01 ce 81 fe ff 7f 00 00 7f d5 81 fe 00 80 ff ff 0f Apr 10 16:53:09 voipmysitesystemd[1]: session-113.scope: Succeeded. Apr 10 16:53:19 voipmysitesystemd[1]: Stopping User Manager for UID 0... Have a nice day! By: Joshua C. Colp (jcolp) 2023-04-11 04:17:19.170-0500 Which opus implementation are you using? The Sangoma provided one? And if you disable opus does the issue persist? By: Pham Huu Phu (jonathanpham) 2023-04-11 05:10:18.492-0500 Dear Joshua, We have been using Asterisk / PJSIP / webRTC. We're use GlobalConnect Trunk. We don't know Sangoma. I thought opus require to use on webRTC? By: Joshua C. Colp (jcolp) 2023-04-11 05:18:35.284-0500 Opus is not required for WebRTC. ULAW and G722 are also supported. How did you get Opus support in Asterisk? Did you select it in "make menuselect"? By: Pham Huu Phu (jonathanpham) 2023-04-11 05:45:23.970-0500 Dear Joshua, We find that We don't select 'opus' in menu select in Asterisk 20.2.0. By: Joshua C. Colp (jcolp) 2023-04-11 05:51:38.049-0500 What is the output of "module show like opus" in the Asterisk console? By: Pham Huu Phu (jonathanpham) 2023-04-11 20:44:58.185-0500 Dear Joshua, I run command on server right now voipmysite*CLI> module show like opus Module Description Use Count Status Support Level codec_opus.so OPUS Coder/Decoder 0 Running extended format_ogg_opus.so OGG/Opus audio 0 Running core res_format_attr_opus.so Opus Format Attribute Module 1 Running core 3 modules loaded From previous version, We built with Opus before we switch to webRTC. Should I disabled Opus to check? Have a nice day! By: Joshua C. Colp (jcolp) 2023-04-12 03:50:41.901-0500 Yes, not using Opus would help to further isolate the underlying issue. By: Pham Huu Phu (jonathanpham) 2023-04-14 02:14:23.219-0500 Thank Joshua, I will try and isolate the problem with codecs. Have a nice day. By: Pham Huu Phu (jonathanpham) 2023-04-25 05:56:39.925-0500 Dear Joshua, We disabled opus. Today We got a crashed again. We attachment files / core and debug file (2023-04-25-*) Event crashed happen Apr 25 12:29:27 voipmysitekernel: show_signal_msg: 8 callbacks suppressed. We appreciate your help. Have a nice day By: Joshua C. Colp (jcolp) 2023-04-25 05:59:36.363-0500 The log shows that opus is still in use, and is still being transcoded. |