[asterisk-ss7] chan_ss7

[Teodor Georgiev]

 Why don't you speak with the chan_ss7 developers?
It is a matter of a single working day for them to implement the "SUS" 
functionality.

On Wednesday 01 March 2006 11:03, leonimar cape wrote:
> Thanks for the replies, I am now currently
> coordinating with the DMS guy if they can disable the
> SUS message. As for the ring back tone I will be
> posting my debug.
>
> Thanks again...
>
> Cheers!
>
> --- Teodor Georgiev <tgeorgiev at is-bg.net> wrote:
> > Thank you for your suggestion. I will spend a lot of
> > time reading to
> > understand them.
> >
> > Now serious... The guy has a direct side-to-side
> > connection.
> > There is no STP involved at all.
> >
> > The chan_ss7 implementation does not support SUS
> > messages. You might look at
> > the source code - there is no definition of the SUS
> > message type. Therefore
> > an Asterisk with chan_ss7 does not know how to react
> > upon receiving an
> > incoming SUS message.
> >
> > On Wednesday 01 March 2006 10:30, Tim Danner wrote:
> > > No kidding, also need "so" to understand adj to
> >
> > non adj routing
> >
> > > commands, l3 for signal management, l4 isup or
> >
> > sccp (tcap) for routing.
> >
> > > -----Original Message-----
> > > From: Teodor Georgiev [mailto:tgeorgiev at is-bg.net]
> > > Sent: Wednesday, March 01, 2006 12:24 AM
> > > To: asterisk-ss7 at lists.digium.com
> > > Subject: Re: [asterisk-ss7] chan_ss7
> > >
> > >
> > >
> > > Here are some fresh news for you --> The chan_ss7
> >
> > does not support the
> >
> > > "SUS"
> > > message. Just checked in isup.h :)
> > >
> > > On Wednesday 01 March 2006 03:18, leonimar cape
> >
> > wrote:
> > > > Hi Group,
> > > >
> > > > I want to asked if someone has successfully
> > > > interconnected asterisk to a telco switch via
> >
> > SS7
> >
> > > > using the chan_ss7? I was able set-up it
> >
> > successfully
> >
> > > > and interconnect it with a DMS Nortel switch.
> >
> > Also the
> >
> > > > quality is indeed perfect. But may issue is on
> >
> > the
> >
> > > > billing (CDR). In a call set up wherein the
> >
> > caller is
> >
> > > > on the asterisk side and the called party is on
> >
> > the
> >
> > > > DMS side, my circuit is not being release even
> >
> > though
> >
> > > > the called party already hungs-up the phone. I
> >
> > know
> >
> > > > that SUS will be send by the DMS to the
> >
> > asterisk, but
> >
> > > > SUS has not been included yet so it seems that
> > > > asterisk dont know that to do next. Circuit is
> >
> > only
> >
> > > > being release if the calling party hungs-up the
> >
> > phone.
> >
> > > > Also, another issue that I have notice is that
> >
> > there
> >
> > > > is no ring back tone receive by the caller. This
> >
> > is
> >
> > > > only on a call setup where the caller is on the
> >
> > DMS
> >
> > > > side and the called party is on the asterisk
> >
> > side. I
> >
> > > > try to apply the RBT patch posted on the
> >
> > voip-info but
> >
> > > > it wasnt successfull.
> > > >
> > > > Any help and suggestion will be greatly
> >
> > appreciated.
> >
> > > > Thanks in advance.
> > > >
> > > > Leonimar Cape
> >
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