[asterisk-ss7] chan_ss7

[leonimar cape]

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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