[asterisk-bugs] [JIRA] (ASTERISK-22853) SIP call hangup during conversation randomly

[Cyril CONSTANTIN (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22853?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=211915#comment-211915 ] 

Cyril CONSTANTIN commented on ASTERISK-22853:
---------------------------------------------

Hi,

I made a test by stopping using MixMonitor for call distributed in queue during two days, and calls for others users were not disconnected anymore. After this test I've activated back MixMonitor on calls queued and all my users faced again the same issue and have their calls disconnected randomly. I just have again tried to deactivate MixMonitor today then since 6 hours all my SIP users are happy, I asked them if issue appears again but it didn't. Generally this problem occurs several times per hour and per user, so I'm getting a feedback quickly from them.

See below what I'm using for MixMonitor, there is nothing special I'm just passing two ARGS to my perl script
{noformat}same => n,Set(MIXMONITOR_FILENAME=/var/www/recording/${STRFTIME(${EPOCH},,%Y%m%d-%H-%M-%S)}-${var})
same => n,MixMonitor(${MIXMONITOR_FILENAME}.wav,v(4),/usr/local/sbin/getagentID.pl ^{UNIQUEID} ^{MIXMONITOR_FILENAME})
{noformat}

Base on my different test I'm really suspecting that issue is coming from MixMonitor function which hangup calls.

Best Regards
                
> SIP call hangup during conversation randomly
> --------------------------------------------
>
>                 Key: ASTERISK-22853
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22853
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.24.0
>         Environment: Debian 6
>            Reporter: Cyril CONSTANTIN
>            Severity: Critical
>         Attachments: filtered - Asteriskooh323 to AvayaClan.rar, logs.rar, tcpdump sip call.rar
>
>
> Hi Team,
> I'm facing a random issue, when SIP user make call through ooh323 their call are hangup during conversation randomly, there is no specific duration where call hangup, it doesn't affect all calls but some per day per SIP user.
> System was working for several month without issue but since I have created a team working in queue and making lot of outbound calls with MIXMONITOR recording them it looks that they have started to get this issue, I'm not sure if it's related but issue started since I have introduced it apparently.
> I was working with version 1.8.15.1 and then I have upgraded to 1.8.24.0 but it didn't resolved the issue.
> I got two example this morning where two SIP user (not user from queue) where making outbound calls and were cut at the same time:
> 1st SIP user was calling number 0760399590
> 2nd SIP user was calling number 0659579568
> Asterisk doesn't crash, SIP calls are just dropped, peer still registered to Asterisk but can't make any outbound calls for several second.
> I have joined all needed traces and below a link with full tcpdump traces:
> http://myaccount.dropsend.com/file/3bdc2d347254db5b
> Let me know if you need anything else.
> Best Regards

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