[asterisk-bugs] [JIRA] (ASTERISK-27985) chan_pjsip will not respond to INVITES when cel_aggregation_topic queue length is > 0
Richard Mudgett (JIRA)
noreply at issues.asterisk.org
Wed Jul 25 06:20:54 CDT 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27985?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Richard Mudgett updated ASTERISK-27985:
---------------------------------------
Description:
We've had sporadic reports of AGIs causing audio glitches on heavy systems (which we believe is related to 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($[ ${COUNT} > 1000 ]?toomany)
same => n,Gosub(testtrigger)
same => n,Set(COUNT=$[ ${COUNT} + 1 ])
same => n,Goto(loop)
same => n(toomany),Hangup
same => n(testtrigger),Dial(SIP/8675309 at 127.0.0.1:5160,1,g)
same => n,Return
{code}
The above dialplan simulates a call entering the 'from-pstn' context of FreePBX, which then does - after the standard dialplan - a Hangup.
This causes Asterisk to create about 20-30 calls per second (but does not trigger the audio issue, unfortunately). It does however trigger a new issue where PJSIP channel INVITES are ignored until the cel_aggregation_topic queue is empty:
{code}
Processor Processed In Queue Max Depth Low water High water
... skipped lines ...
subm:cdr_engine-00000003 5780891 0 2347 4500 5000
subm:cel_aggregation_topic-00000006 4968215 786689 2961396 2700 3000
subm:endpoint_topic_all-cached-00000008 794 0 13 450 500
subm:endpoint_topic_all-cached-0000005a 778 0 13 450 500
subm:manager_topic-00000007 5962603 0 344 2700 3000
{code}
While the queue is being processed, pjsip calls are ignored. chan_sip calls are answered and processed correctly, increasing the queue size.
was:
We've had sporadic reports of AGIs causing audio glitches on heavy systems (which we believe is related to https://issues.asterisk.org/jira/browse/ASTERISK-26257 but is NOT limited to confbridge, MOH and Playback/Background is also affected).
{code}
exten => 998,1,Answer
same => n,Dial(Local/999 at from-internal-custom/n,300,gm(default))
same => n,Playback(beep)
same => n,Goto(1)
exten => 999,1,Set(COUNT=0)
same => n(loop),GotoIf($[ ${COUNT} > 1000 ]?toomany)
same => n,Gosub(testtrigger)
same => n,Set(COUNT=$[ ${COUNT} + 1 ])
same => n,Goto(loop)
same => n(toomany),Hangup
same => n(testtrigger),Dial(SIP/8675309 at 127.0.0.1:5160,1,g)
same => n,Return
{code}
The above dialplan simulates a call entering the 'from-pstn' context of FreePBX, which then does - after the standard dialplan - a Hangup.
This causes Asterisk to create about 20-30 calls per second (but does not trigger the audio issue, unfortunately). It does however trigger a new issue where PJSIP channel INVITES are ignored until the cel_aggregation_topic queue is empty:
{code}
Processor Processed In Queue Max Depth Low water High water
... skipped lines ...
subm:cdr_engine-00000003 5780891 0 2347 4500 5000
subm:cel_aggregation_topic-00000006 4968215 786689 2961396 2700 3000
subm:endpoint_topic_all-cached-00000008 794 0 13 450 500
subm:endpoint_topic_all-cached-0000005a 778 0 13 450 500
subm:manager_topic-00000007 5962603 0 344 2700 3000
{code}
While the queue is being processed, pjsip calls are ignored. chan_sip calls are answered and processed correctly, increasing the queue size.
> chan_pjsip will not respond to INVITES when cel_aggregation_topic queue length is > 0
> -------------------------------------------------------------------------------------
>
> Key: ASTERISK-27985
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27985
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: CEL/cel_odbc
> Affects Versions: 15.4.1, 13.22.0
> Environment: FreePBX 14, SNG7
> Reporter: xrobau
> Labels: pjsip
> Attachments: pjsip-ignored.txt
>
>
> We've had sporadic reports of AGIs causing audio glitches on heavy systems (which we believe is related to ASTERISK-26257 but is NOT limited to confbridge, MOH and Playback/Background is also affected).
> {code}
> exten => 998,1,Answer
> same => n,Dial(Local/999 at from-internal-custom/n,300,gm(default))
> same => n,Playback(beep)
> same => n,Goto(1)
> exten => 999,1,Set(COUNT=0)
> same => n(loop),GotoIf($[ ${COUNT} > 1000 ]?toomany)
> same => n,Gosub(testtrigger)
> same => n,Set(COUNT=$[ ${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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