[asterisk-ss7] chan_ss7

[Anders Baekgaard]

Hi

The suspend/resume handling in chan_ss7 is currently under development, and a 
new release is planned within a couple of days. Stay tuned...

Best regards,
Anders Baekgaard

On Wednesday 01 March 2006 16:19, Teodor Georgiev wrote:
>  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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