[asterisk-ss7] Re: Release cause problem

[Peter Demeter]

As I wrote in my first post:
IP phone calls PSTN -> IP side releases call

My Dial command is "Dial(SS7/${EXTEN},120,g)", after this I have a Hangup.
If the IP phone releases the call then chan_ss7 disconnects the call with
release cause 41 because it receives 0 from Asterisk and then Hangup is
executed, but it's not affecting the release cause because it was already
sent to the PSTN switch.

What else can be specified in the Dial command which might affect the
result?

On 11/22/06, Anton <anton.vazir at gmail.com> wrote:
>
> if calling party answers the call, next dialplan step is not
> executed, unles you set appropriate option in Dial() - so
> it will not pass you HangupCause.
>
> So could you write more specific what side call what what
> side in what condition returns whhat code and what
> causecode and where you receive
>
> On 22 November 2006 17:24, Peter Demeter wrote:
> > You mean Hangup in extensions.conf? I tried.  It is
> > executed after chan_ss7 disconnects the call so no
> > effect.
> >
> > Thank you anyway.
> >
> > Regards,
> > Peter
> >
> > Anton anton.vazir at gmail.com
> > Wed Nov 22 04:18:08 MST 2006
> >
> > Hangup(${DISCONNECTCAUSE}) or made a conditional
> > Hangup(CAUSECODE) if returned causecodes are non-standard
> >
> > On 22 November 2006 15:49, Peter Demeter wrote:
> > > Hi,
> > >
> > > I have a problem with chan_ss7 & Asterisk.
> > >
> > > If an IP Phone (SIP) calls a PSTN number and the call
> > > is terminated by the calling party chan_ss7 won't (?)
> > > recieve a release cause from Asterisk.  In my case this
> > > means that the release cause will be 41 (Temporary
> > > Failure) in the connecting EWSD switch instead of 16
> > > (Normal Call Clearing).
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