[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=222935#comment-222935 ] 

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

by the way, I just found that I have debug log, as I said it changes tones, so I don't sure it is right.



> 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: h323_log, 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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