[Home]

Summary:ASTERISK-26640: app_queue: Asterisk crashes when dialplan execution goes to queue application
Reporter:Yuriy Topin (yuratop)Labels:
Date Opened:2016-12-05 03:27:21.000-0600Date Closed:2016-12-05 09:45:00.000-0600
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_queue
Versions:13.8.2 Frequency of
Occurrence		Occasional
Related
Issues:
Environment:Linux pbx.host 4.4.0-31-generic #50~14.04.1-Ubuntu SMP Wed Jul 13 01:07:32 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux Attachments:
Description:When dialplan goes to Queue application asterisk crashes (13.8.2 cert2 version).
Comments:By: Asterisk Team (asteriskteam) 2016-12-05 03:27:22.751-0600

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: Yuriy Topin (yuratop) 2016-12-05 03:32:29.398-0600

[Edit by Rusty - removed large inline debug. Please do not attach large amounts of debug into a comment field.]
By: Yuriy Topin (yuratop) 2016-12-05 03:36:08.590-0600

Last log lines in the dialplan.


[Dec  5 10:06:31] DEBUG[23391][C-00000004]: app_queue.c:3710 join_queue: Queue 'call1' Join, Channel 'SIP/380674868466-00000004', Position '1'
[Dec  5 10:06:31] DEBUG[23179]: app_queue.c:2470 device_state_cb: Device 'Queue:call1' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
   -- Started music on hold, class 'default', on channel 'SIP/380674868466-00000004'
[Dec  5 10:06:31] DEBUG[23391][C-00000004]: channel.c:3424 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Dec  5 10:06:31] DEBUG[23391][C-00000004]: channel.c:4905 ast_prod: Prodding channel 'SIP/380674868466-00000004'
[Dec  5 10:06:31] DEBUG[23391][C-00000004]: res_rtp_asterisk.c:3466 ast_rtp_write: Received frame with no data for RTP instance '0x7f7570075178' so dropping frame
[Dec  5 10:06:31] DEBUG[23391][C-00000004]: app_queue.c:5249 is_our_turn: There is 1 available member.
[Dec  5 10:06:31] DEBUG[23391][C-00000004]: app_queue.c:5264 is_our_turn: It's our turn (SIP/380674868466-00000004).

Then asterisk stops.
By: Joshua C. Colp (jcolp) 2016-12-05 09:45:01.009-0600

Asterisk Certified only receives fixes as a result of issues reported by Digium customers who have a suitable agreement for support on it. If you are one of these customers please contact Digium Technical Support to go through the process.

If you are not a customer I would suggest trying the latest version of certified and also the mainline version of Asterisk. If in the latest mainline Asterisk you are still experiencing a problem please open a new issue and attach a backtrace.