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

Dmitry Melekhov (JIRA) noreply at issues.asterisk.org
Tue Oct 7 23:31:29 CDT 2014


Dmitry Melekhov created ASTERISK-24400:
------------------------------------------

             Summary: 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
            Severity: Minor


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:

exten => 5880,1,SET(FAXOPT(t38gateway)=yes)
exten => 5880,n,Progress
exten => 5880,n,Dial(DAHDI/g1/${EXTEN})
exten => 5880,n,Hangup

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:

kt-ta100---asterisk--cisco3845--ts004-avaya

I'm avaya user :-)

So, what I see is on asterisk:

    -- Called DAHDI/g1/5880
    -- Span 1: Channel 0/2 got hangup, cause 17

on cisco:

Oct  8 03:27:58.138: //1355041/EE3142978C7C/CCAPI/ccCallDisconnect:
   Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38)

I placed call through another asterisk, which is connected to ts004, over ISDN, call is over ISDN (back to ts004, etc..) , and see there:

    -- Span 1: Channel 0/3 got hangup request, cause 38
    -- DAHDI/i1/5880-21aa is circuit-busy

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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