[asterisk-bugs] [JIRA] (ASTERISK-24400) ooh323 sends wrong hangup code

[Dmitry Melekhov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24400?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223046#comment-223046 ] 

Dmitry Melekhov commented on ASTERISK-24400:
--------------------------------------------

Hello!

With patched asterisk I do not get 38 tone with avaya anymore.
And yes, it returns 16 (i.e. hangup), not 17 (user busy) for some reason:

    -- DAHDI/i1/7500-a is proceeding passing it to OOH323/192.168.22.253-9
    -- Span 1: Channel 0/2 got hangup request, cause 16
    -- Hungup 'DAHDI/i1/7500-a'
    -- No one is available to answer at this time (1:0/0/0)

I have to ask avaya admin about this, because another definity works as expected, and yes, asterisk there has, oops, definitely had :-) the same issue, nobody complained though, this is because most calls there are between asterisks.

Thank you very much!

Is this fix stable or you want to, let's say, stabilize it? In last case I'm ready to test.

Thank you!


> ooh323 sends wrong hangup code
> ------------------------------
>
>                 Key: ASTERISK-24400
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24400
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/chan_ooh323
>    Affects Versions: 11.10.2
>         Environment: centos x86-64
>            Reporter: Dmitry Melekhov
>            Assignee: Alexander Anikin
>            Severity: Minor
>         Attachments: 24400-test-2.patch, 24400-test-3.patch, 24400-test-4.patch, 24400-test.patch, cisco-debug.dmp, cisco_log, cisco_log_tcp, cisco_log_tcp_progress, cisco-nodebug.dmp, debug.dump, debug-noprogress.dump, h323_log, h323_log, h323_log, h323_log, h323_log, h323_log, nodebug.dump, nodebug-noprogress.dump, ooh323.conf
>
>
> Hello!
> We installed yet another asterisk which works as voip gateway between panasonic kx-ta 100 and our h323 network.
> Connection between asterisk and panasonic is ISDN PRI, namely qsig.
> For some reason we have problems with generating ring back tone from panasonic , so I added Progress, so it looks like:
> {noformat}
> exten => 5880,1,SET(FAXOPT(t38gateway)=yes)
> exten => 5880,n,Progress
> exten => 5880,n,Dial(DAHDI/g1/${EXTEN})
> exten => 5880,n,Hangup
> {noformat}
> but, in this case we get wrong hangup tone, like not user busy, but network congestion.
> If there is no Progress, then no ringback, but tone is right.
> Really, whole connection scheme is:
> {noformat}
> kt-ta100---asterisk--cisco3845--ts004-avaya
> {noformat}
> I'm avaya user :-)
> So, what I see is on asterisk:
> {noformat}
>     -- Called DAHDI/g1/5880
>     -- Span 1: Channel 0/2 got hangup, cause 17
> {noformat}
> on cisco:
> {noformat}
> Oct  8 03:27:58.138: //1355041/EE3142978C7C/CCAPI/ccCallDisconnect:
>    Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)
> {noformat}
> I placed call through another asterisk, which is connected to ts004, over ISDN, call is over ISDN (back to ts004, etc..) , and see there:
> {noformat}
>     -- Span 1: Channel 0/3 got hangup request, cause 38
>     -- DAHDI/i1/5880-21aa is circuit-busy
> {noformat}
> I don't completely sure how to collect all data, because  if I turn on debug or trace then I got different behaviour, namely I don't hear ringback tone even if Progress is in place.
> So I need help from developer to collect debug info.
> I guess that problem is in ooh323 because if I use sip to call asterisk from cisco3845 , then I get user busy tone.
> Thank you!



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