[asterisk-bugs] [JIRA] (ASTERISK-30112) Call recording truncated without dropping call.

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30112?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259509#comment-259509 ] 

Asterisk Team commented on ASTERISK-30112:
------------------------------------------

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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].

> Call  recording truncated without dropping call.
> ------------------------------------------------
>
>                 Key: ASTERISK-30112
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30112
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_record
>    Affects Versions: 18.10.0
>         Environment: Ubuntu 20.04.03 LTS
> Kernel: Linux 5.4.0-100-generic
> Asterisk 18.10.0 installed from source using tar.gz file.
> Memory : 8 GB  swap 2GB
> Cpu Count: 2
> Cpu Model : Intel(R) Xeon(R) CPU E5-2630 v3 @ 2.40GHz
>            Reporter: e-telequote.com
>            Severity: Major
>
> We are facing this issue where calls recording suddenly truncated without disconnecting the calls. CPU usage at that point in time sometimes well under 80%. 
>       We are using ARI application to generate outbound calls. Two legs were bridged together and then Recorder/ARI channel used to record the audio in the bridge. Also recording happens directly in file in /var/spool/asterisk/recording/filename. Means not in temp file in temo directory. After looking into log files finds this.
> VERBOSE[354962] bridge.c: Bridge d8f9e148-122c-4a70-a26a-5c03385bf581: switching from simple_bridge technology to softmix
> WARNING[354964][C-00000d91] app.c: No audio available on Recorder/ARI-000006ee;1??
> WARNING[354962] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-000006ee;1
> WARNING[354962] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-000006ee;1
> WARNING[354962] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-000006ee;1
> WARNING[354962] channel.c: Exceptionally long voice queue length queuing to Recorder/ARI-000006ee;1
> VERBOSE[354962] bridge_channel.c: Channel Recorder/ARI-000006ee;2 left 'softmix' stasis-bridge <d8f9e148-122c-4a70-a26a-5c03385bf581>
> VERBOSE[354962] bridge_channel.c: Channel Recorder/ARI-000006ee;2 left 'softmix' stasis-bridge <d8f9e148-122c-4a70-a26a-5c03385bf581>
> VERBOSE[354962] bridge.c: Bridge d8f9e148-122c-4a70-a26a-5c03385bf581: switching from softmix technology to native_rtp
> VERBOSE[354952] bridge_channel.c: Channel SIP/172.24.98.14-00001c2b left 'native_rtp' stasis-bridge <d8f9e148-122c-4a70-a26a-5c03385bf581>
> VERBOSE[354952] bridge.c: Bridge d8f9e148-122c-4a70-a26a-5c03385bf581: switching from native_rtp technology to simple_bridge
> VERBOSE[354951][C-00000d91] bridge_channel.c: Channel SIP/outbound-00001c21 left 'simple_bridge' stasis-bridge <d8f9e148-122c-4a70-a26a-5c03385bf581>
> Sophos anti-virs and TM Deep security agent are running on the server. So trying to figure out what cause this issue. Also need guidelines and help to further advance our investigation in right direction to locate/resolve the issue.



--
This message was sent by Atlassian JIRA
(v6.2#6252)