Hi,
it is not a Telco standart, however some countries national SS7 implementation
uses SUSpend in the following case:
A call, dropped by the calling party is released in that manner:
A B
REL -->
<-- RLC
However, calls dropped by the called party are released in the following
manner:
A B
<-- SUS
REL -->
<-- RLC
About your ringback, with the receiving of the ACM message, the PSTN switches
should establish a voice circuit where the ringback is to be sent.
You better make and post an ISUP debug here :)
On Wednesday 01 March 2006 09:34, leonimar cape wrote:
> I think it is a telco standard, and it is part of the
> switch to switch testing. I will asked them if they
> can omit it. How about my 2nd issue wherein the ring
> backtone did not being received by the calling party?
> Any idea on this?
>
> --- Teodor Georgiev <tgeorgiev@xxxxxxxxx> wrote:
> > Well,
> >
> > the usual way to end a call is REL-RLC. Why the DMS
> > is sending a SUS?
> >
> >
> > 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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