[asterisk-dev] Wrong SIP to SIP SIP Cause mapping

[Pavel Troller]

Hi!

Oh, Mamma mia, Alex is right!!!!
  I've just tested a void public number through PRI and I've got CC=1 - 
Unallocated number.
  Then I dialled the same number through SIP trunk and I've got CC=34,
but on the SIP the response code was really 404 Not Found, which is
equivalent to 1 - Unallocated number. 
  Something is really broken here! Of course I'm paying attention to
$HANGUPCAUSE and I'm handling it in my own intercept scripts, and they
are proved OK, so the problem must be in wrong $HANGUPCAUSE contents.
  It's 1.8 branch.
  I hope I will find the cause soon and I'll post it here.
    With regards, Pavel.

> Hi,
> 
> That is the problem, asterisk receive 404 but hangupcause is 34 and not 1.
> When i try with version 1.8.8.0 or before, I have the hangupcause 1 and with all newer version, I have hangupcause 34.
> 
> alex
> 
> 
> 
> ----- Original Message -----
> > From: "Ryan Mitchell" <rjm at tcl.net>
> > To: "Asterisk Developers Mailing List" <asterisk-dev at lists.digium.com>
> > Sent: Friday, May 25, 2012 5:15:38 PM
> > Subject: Re: [asterisk-dev] Wrong SIP to SIP SIP Cause mapping
> > What is $HANGUPCAUSE set to after the Dial() ?
> > 
> > 
> > Asterisk is a b2bua, not a proxy, as you know. Often in my scripts I
> > am paying attention to $HANGUPCAUSE and calling Hangup() with explicit
> > arguments.
> > 
> > 
> > Ryan
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > On Fri, May 25, 2012 at 3:28 PM, alexandre Moutot <
> > a.moutot at alphalink.fr > wrote:
> > 
> > 
> > It is ... What do you need to believe me ?
> > 
> > 
> > 
> > ----- Original Message -----
> > > From: "Olle E. Johansson" < oej at edvina.net >
> > > To: "Asterisk Developers Mailing List" <
> > > asterisk-dev at lists.digium.com >
> > > Sent: Friday, May 25, 2012 2:55:59 PM
> > > Subject: Re: [asterisk-dev] Wrong SIP to SIP SIP Cause mapping
> > > 25 maj 2012 kl. 14:17 skrev alexandre Moutot:
> > >
> > > > Hello,
> > > >
> > > >
> > > > I'm using asterisk v1.8 with a standard scenario, A Sip call from
> > > > A
> > > > to B through asterisk :
> > > >
> > > > A --SIP--> ASTERISK --SIP--> B
> > > >
> > > > The asterisk extension is :
> > > > exten => _X.,1,Dial(SIP/B/${EXTEN},600)
> > > > exten => _X.,n,Hangup()
> > > >
> > > > When B send a 404 back to the asterisk, the asterisk sends a 503
> > > > to
> > > > A. It is the same with 403 and some others erroc code.
> > > > I think it should send back to A the same error code.
> > > >
> > > > I have done tests with some versions:
> > > > - 1.8.11.x : wrong sip cause mapping
> > > > - 1.8.12.0 : wrong sip cause mapping
> > > > - 1.8.13.0rc1 : wrong sip cause mapping
> > > > - 1.10.3 : wrong sip cause mapping
> > > > - 1.8.8.0 : works good
> > > >
> > > > Do i do something wrong or should i open a bug ?
> > >
> > > We are not always sending the very same code, but a 4xx class code
> > > should not be converted to a 5xx class.
> > >
> > > /O
> > > --
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> > 
> > 
> > --
> > Ryan Mitchell < rjm at tcl.net >
> > Telecom Logic, LLC
> > 
> > 
> > 
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