[asterisk-bugs] [JIRA] (ASTERISK-27985) chan_pjsip will not respond to INVITES when cel_aggregation_topic queue length is > 0

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27985?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=244239#comment-244239 ] 

Asterisk Team commented on ASTERISK-27985:
------------------------------------------

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.

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> chan_pjsip will not respond to INVITES when cel_aggregation_topic queue length is > 0
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27985
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27985
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: CEL/cel_odbc
>    Affects Versions: 13.22.0
>         Environment: FreePBX 14, SNG7
>            Reporter: xrobau
>              Labels: pjsip
>         Attachments: pjsip-ignored.txt
>
>
> We've had sporadic reports of AGIs causing audio glitches on heavy systems (which we believe is related to https://issues.asterisk.org/jira/browse/ASTERISK-26257 but is NOT limited to confbridge, MOH and Playback/Background is also affected).
> {code}
> exten => 998,1,Answer
>  same => n,Dial(Local/999 at from-internal-custom/n,300,gm(default))
>  same => n,Playback(beep)
>  same => n,Goto(1)
> exten => 999,1,Set(COUNT=0)
>  same => n(loop),GotoIf($[ ${COUNT} > 1000 ]?toomany)
>  same => n,Gosub(testtrigger)
>  same => n,Set(COUNT=$[ ${COUNT} + 1 ])
>  same => n,Goto(loop)
>  same => n(toomany),Hangup
>  same => n(testtrigger),Dial(SIP/8675309 at 127.0.0.1:5160,1,g)
>  same => n,Return
> {code}
> The above dialplan simulates a call entering the 'from-pstn' context of FreePBX, which then does - after the standard dialplan - a Hangup. 
> This causes Asterisk to create about 20-30 calls per second (but does not trigger the audio issue, unfortunately).  It does however trigger a new issue where PJSIP channel INVITES are ignored until the cel_aggregation_topic queue is empty:
> {code}
> Processor                                      Processed   In Queue  Max Depth  Low water High water
> ... skipped lines ...
> subm:cdr_engine-00000003                         5780891          0       2347       4500       5000
> subm:cel_aggregation_topic-00000006              4968215     786689    2961396       2700       3000
> subm:endpoint_topic_all-cached-00000008              794          0         13        450        500
> subm:endpoint_topic_all-cached-0000005a              778          0         13        450        500
> subm:manager_topic-00000007                      5962603          0        344       2700       3000
> {code}
> While the queue is being processed, pjsip calls are ignored.  chan_sip calls are answered and processed correctly, increasing the queue size.



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