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

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22853?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212884#comment-212884 ] 

Rusty Newton edited comment on ASTERISK-22853 at 1/3/14 6:36 PM:
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Thanks. Since the frequency of the issue has greatly reduced and you have moved from 1.8 to 11, I'd say give it more time as you could encounter new issues just from moving across two LTS releases. Let us know in a couple weeks.
                
      was (Author: rnewton):
    Thanks. Since the frequency of the issue has greatly occurred and you have moved from 1.8 to 11, I'd say give it more time as you could encounter new issues just from moving across two LTS releases. Let us know in a couple weeks.
                  
> SIP call hangup randomly during conversation due to MixMonitor
> --------------------------------------------------------------
>
>                 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
>            Assignee: 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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