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

Ryan Mitchell rjm at tcl.net
Fri May 25 10:15:38 CDT 2012


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
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>



-- 
Ryan Mitchell <rjm at tcl.net>
Telecom Logic, LLC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120525/566120aa/attachment-0001.htm>


More information about the asterisk-dev mailing list