[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=212188#comment-212188 ] 

Rusty Newton commented on ASTERISK-22853:
-----------------------------------------

To know for sure what is going on here, we'll need full pcaps of working calls (where mixmonitor was used on their channels as well)

As far as I can tell from your current pcaps, without going into various oddities that are probably unrelated, it looks like the phones are sending Asterisk a bye to hang up the call.

If Asterisk is doing something bad that results in the phone sending the BYE... I don't know, but possibly comparing audio between pcaps of working and non working calls may help.

Do the users on the phone report any audio abnormalities before the call is hung up?


                
> 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
>            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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