| Summary: | ASTERISK-26663: Chan_sip deadlock with local channel and audiohooks | ||||
| Reporter: | Ruddy G (ruddy) | Labels: | |||
| Date Opened: | 2016-12-15 12:36:18.000-0600 | Date Closed: | 2016-12-21 10:29:59.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | |||
| Versions: | 11.14.0 11.25.1 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ( 0) asterisk-11.14.backtrace ( 1) asterisk-11.25.backtrace | |||
| Description: | Hi,
I discovered what creates a deadlock in asterisk and prevent any new sip registration to occur. There are two scenarios. Scenario 1: A local channel calls an application that installs audiohooks on it before calling a SIP user. At the end of the call, we have a deadlock. Scenario 2: A SIP user calls a PBX which dial a Local channel with an announcement. That local channel dials another SIP user. This last user answers the phone and transfers it before the announcement is completed. Scenario 1: Create a p.call file wih a local channel Channel: Local/51400000000@my-inbound Application: SayAlpha Data: 123456789 Inside [my-inbound] context, have the channel create an audiohook and then dial a SIP trunk. At the end of the call, the PBX SIP module is deadlocked. No new registration is allowed. Here are the relevant threads backtraces: Scenario 2: Caller A calls from outside and reach context [my-inbound] Inside such context, we do dial a local context with announcement: Dial(Local/000@my-local-context,A(myaudio)) Inside the [my-local-context], simply call SIP user B [my-local-context] exten => 000,1,Dial(SIP/userb,30) So, user B answers the phone. Before his myaudio.gsm announcement is completed, he transfers the call to another location. Basically, B just transfer his own local channel instead of SIP/userA A deadlock occurs with the same backtraces. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2016-12-15 12:36:19.616-0600 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: Ruddy G (ruddy) 2016-12-15 13:54:23.036-0600 I'm having now the same issue on 11.14.0 and on 11.25.1 By: Ruddy G (ruddy) 2016-12-15 14:25:34.857-0600 See attached 11.25 bug By: Rusty Newton (rnewton) 2016-12-20 12:41:16.032-0600 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: Ruddy G (ruddy) 2016-12-21 09:07:15.813-0600 Bug backtrace By: Rusty Newton (rnewton) 2016-12-21 10:28:39.541-0600 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 By: Rusty Newton (rnewton) 2016-12-21 10:29:59.625-0600 Please don't open this issue again for 11. It appears we closed your previous nearly identical issue for the same reason. If you can reproduce an issue on a branch that is supported for bug fixes then feel free to open a new issue. Please look at the Asterisk versions page that is linked. | ||||