| Summary: | ASTERISK-26372: dial: Crash when removing dial masquerade datastore from peer | ||||
| Reporter: | Jaco van Niekerk (faqterson) | Labels: | |||
| Date Opened: | 2016-09-14 03:15:11 | Date Closed: | 2016-09-15 06:18:46 | ||
| Priority: | Major | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | Applications/app_dial | ||
| Versions: | 13.10.0 | Frequency of Occurrence | |||
| Related Issues: |
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| Environment: | Centos 5.11 32Gig Memory Intel(R) Xeon(R) CPU E5-2630 v2 @ 2.60GHz Software RAID | Attachments: | ( 0) coredump.txt ( 1) coredump2.txt ( 2) CoredumpFull.txt | ||
| Description: | Experiencing intermittent asterisk restarts.
Core dumps attached | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2016-09-14 03:15:12.146-0500 We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum. The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors. If this issue is actually a bug please use the Bug issue type instead. Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Asterisk Team (asteriskteam) 2016-09-14 03:15:13.190-0500 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: Jaco van Niekerk (faqterson) 2016-09-14 03:18:23.886-0500 I will like to confirm that this is not dial plan related? By: Asterisk Team (asteriskteam) 2016-09-14 03:18:23.993-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Rusty Newton (rnewton) 2016-09-14 09:49:55.336-0500 Thanks for the report! In order to maximize efficiently, there are project guidelines for how to report issues. Please read through the Asterisk Issue Guidelines [1]. After reading the guidelines, please clean up this issue so that bug marshals can more easily help you. In particular: 1. Don't post extensive debug or logs inside the Description or Comment fields. 2. Use the Description field for a description of the issue, referencing *attached* debug with links or notes. 3. Use the Comment fields for discussion regarding the issue. 4. If you need to put a few lines of debug or logs into any field, surround the text with "noformat" tags to help us read it easily. 5. Attach files with a '.txt' extension where possible so that they can be analyzed futher by bug marshals. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Rusty Newton (rnewton) 2016-09-14 17:40:35.273-0500 Jaco thanks for cleaning this up. You marked this as a regression. What version were you using previous to 13.10 where this wasn't happening? Also, you should note that there was a security release since then and that you should upgrade to the latest 13.X release. Do you have any messages or full log files from the crash? For future crashes, can you go ahead and prepare to get a log file with both VERBOSE and DEBUG turned up to 5 or higher. You'll also want to verify the file has WARNING, ERROR and NOTICE turned on. By: Jaco van Niekerk (faqterson) 2016-09-15 00:47:26.257-0500 Yes, its been on going issue. Its been happening randomly on Asterisk 13.10.0 Sep 6 13:04 Sep 6 15:06 Sep 8 10:21 Sep 8 14:55 Sep 9 14:43 Sep 12 07:28 Sep 13 14:20 I have now increased the verbose and Debug to 5. By: Jaco van Niekerk (faqterson) 2016-09-15 00:49:08.641-0500 I am only detecting WARNINGS and one NOTICE at the time it restarted: [2016-09-12 07:28:08] WARNING[24277][C-00000339] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: [2016-09-12 07:28:08] WARNING[24277][C-00000339] ast_expr2.fl: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables [2016-09-12 07:28:23] NOTICE[24277][C-00000339] app_dial.c: privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding By: Jaco van Niekerk (faqterson) 2016-09-15 02:22:18.201-0500 Two of my sites running 600+ extensions and 60+ concurrent calls are experiencing this problem, the core dumps provided are from the first site. The second site I am getting completely different core dumps, mostly related to chan_sip. I have attached a core dump for this site. Sites with 100 extensions and 20 concurrent calls I am not experiencing problems. By: Jaco van Niekerk (faqterson) 2016-09-15 02:25:02.857-0500 Second site experience asterisk restart By: Jaco van Niekerk (faqterson) 2016-09-15 05:31:10.566-0500 Below is the exact information that shows when the asterisk crashes: "noformat" -- Executing [s-OK@macro-trunk:3] Dial("SIP/7007-0000d38a", "IAX2/bitco-phalaborwa/0782164710,300,T") in new stack -- Called IAX2/bitco-phalaborwa/0782164710 -- Call accepted by 41.79.80.11:4569 (format g729) -- Format for call is (g729) -- <SIP/3000-0000d388> Playing 'priv-callerintros/0214221156.slin' (language 'en') -- <SIP/3000-0000d388> Playing 'screen-callee-options.gsm' (language 'en') -- <SIP/3000-0000d388> Playing 'vm-sorry.gsm' (language 'en') -- <SIP/3000-0000d388> Playing 'priv-callerintros/0214221156.slin' (language 'en') -- <SIP/3000-0000d388> Playing 'screen-callee-options.gsm' (language 'en') -- <SIP/3000-0000d388> Playing 'vm-sorry.gsm' (language 'en') -- <SIP/3000-0000d388> Playing 'priv-callerintros/0214221156.slin' (language 'en') -- <SIP/3000-0000d388> Playing 'screen-callee-options.gsm' (language 'en') -- <SIP/3000-0000d388> Playing 'vm-sorry.gsm' (language 'en') [2016-09-15 12:27:50] NOTICE[12768][C-000083ae]: app_dial.c:1858 do_privacy: privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding "noformat" By: Jaco van Niekerk (faqterson) 2016-09-15 05:52:24.442-0500 I suspect I found the problem: -- Executing [s-DIAL@macro-exten:1] Dial("IAX2/bitco-15306", "SIP/3002,300,tTIAX2/workforce-temp") in new stack My dial plan is sending options to the dial application that's wrong. I am going to work on fixing the dial plan now. By: Joshua C. Colp (jcolp) 2016-09-15 06:18:46.512-0500 This is happening because of the privacy feature. There was a crash, recently fixed, when that functionality was used. | ||||