[asterisk-bugs] [JIRA] (ASTERISK-28097) Queue option 'b' + SIP_HEADER make issue

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28097?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=245125#comment-245125 ] 

Asterisk Team commented on ASTERISK-28097:
------------------------------------------

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].

> Queue option 'b' + SIP_HEADER make issue
> ----------------------------------------
>
>                 Key: ASTERISK-28097
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28097
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 16.0.0
>         Environment: CentOS 6 (64bit) + Asterisk 16 
>            Reporter: Ryou, HyunSun
>
> # Dial-Plan
> 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
> # CLI
> asterisk16*CLI> 
>   == Using SIP RTP CoS mark 5
>        > 0x7f29a0010670 -- Strict RTP learning after remote address set to: 19.19.20.45:19572
>     -- Executing [07047318355 at inbound:1] NoOp("SIP/ZABBIX-00000000", "-- 07047318355 From Zabbix --") in new stack
>     -- Executing [07047318355 at inbound:2] UserEvent("SIP/ZABBIX-00000000", "UserChannelInit,inexten: 07047318355") in new stack
>     -- Executing [07047318355 at inbound:3] NoOp("SIP/ZABBIX-00000000", "CALLER=01026694023") in new stack
>     -- Executing [07047318355 at inbound:4] NoOp("SIP/ZABBIX-00000000", "Call-ID=2704ec0f0297d2062ccc6ec2175713d4 at 19.19.20.45:5060") in new stack
>     -- Executing [07047318355 at 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 at QueueDial:1] NoOp("SIP/2001-00000001", "--QueueDial--") in new stack
>     -- Executing [07047318355 at QueueDial:2] NoOp("SIP/2001-00000001", "Channel=SIP/2001-00000001") in new stack
>     -- Executing [07047318355 at QueueDial:3] NoOp("SIP/2001-00000001", "Call-ID=") in new stack
>     -- Executing [07047318355 at 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
> like upper log, asterisk process is shutdonw and make core dump.



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