Summary: | ASTERISK-28097: Queue option 'b' + SIP_HEADER make issue | ||
Reporter: | Ryou, HyunSun (hsunryou) | Labels: | |
Date Opened: | 2018-10-10 21:29:32 | Date Closed: | 2020-01-14 11:14:00.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | . I did not set the category correctly. |
Versions: | 16.0.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | CentOS 6 (64bit) + Asterisk 16 | Attachments: | |
Description: | # Dial-Plan
{noformat} exten => 07047318355,1,Noop(-- ${EXTEN} From Zabbix --) same => n,UserEvent(UserChannelInit,inexten: ${EXTEN}) same => n,Noop(CALLER=${CALLERID(num)}) same => n,Noop(Call-ID=${SIP_HEADER(Call-ID)}) same => n,Queue(TestQueue,b(QueueDial,${EXTEN},1)) same => n,Hangup [QueueDial] exten => _070XXX.,1,noop(--QueueDial--) same => n,Noop(Channel=${CHANNEL}) same => n,Noop(Call-ID=${SIP_HEADER(Call-ID)}) same => n,Return {noformat} # CLI {noformat} asterisk16*CLI> == Using SIP RTP CoS mark 5 > 0x7f29a0010670 -- Strict RTP learning after remote address set to: 19.19.20.45:19572 -- Executing [07047318355@inbound:1] NoOp("SIP/ZABBIX-00000000", "-- 07047318355 From Zabbix --") in new stack -- Executing [07047318355@inbound:2] UserEvent("SIP/ZABBIX-00000000", "UserChannelInit,inexten: 07047318355") in new stack -- Executing [07047318355@inbound:3] NoOp("SIP/ZABBIX-00000000", "CALLER=01026694023") in new stack -- Executing [07047318355@inbound:4] NoOp("SIP/ZABBIX-00000000", "Call-ID=2704ec0f0297d2062ccc6ec2175713d4@19.19.20.45:5060") in new stack -- Executing [07047318355@inbound:5] Queue("SIP/ZABBIX-00000000", "TestQueue,b(QueueDial,07047318355,1)") in new stack -- Started music on hold, class 'default', on channel 'SIP/ZABBIX-00000000' == Using SIP RTP CoS mark 5 -- SIP/2001-00000001 Internal Gosub(QueueDial,07047318355,1) start -- Executing [07047318355@QueueDial:1] NoOp("SIP/2001-00000001", "--QueueDial--") in new stack -- Executing [07047318355@QueueDial:2] NoOp("SIP/2001-00000001", "Channel=SIP/2001-00000001") in new stack -- Executing [07047318355@QueueDial:3] NoOp("SIP/2001-00000001", "Call-ID=") in new stack -- Executing [07047318355@QueueDial:4] Return("SIP/2001-00000001", "") in new stack == Spawn extension (inexten, 07047318355, 1) exited non-zero on 'SIP/2001-00000001' -- SIP/2001-00000001 Internal Gosub(QueueDial,07047318355,1) complete GOSUB_RETVAL= -- Called SIP/2001 asterisk16*CLI> Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups {noformat} like upper log, asterisk process is shutdown and make core dump. | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-10-10 21:29:33.801-0500 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) 2018-10-11 05:03:25.035-0500 Thank you for the crash report. However, we need more information to investigate the crash. Please provide: 1. A backtrace generated from a core dump using the instructions provided on the Asterisk wiki [1]. 2. Specific steps taken that lead to the crash. 3. All configuration information necesary to reproduce the crash. Thanks! [1]: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace By: Asterisk Team (asteriskteam) 2018-10-25 12:00:01.025-0500 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 |