[asterisk-ss7] chan_ss7

[Tim Danner]

Are we talking about when IP may be in the middle?

-----Original Message-----
From: leonimar cape [mailto:leo_mac_ph at yahoo.com] 
Sent: Tuesday, February 28, 2006 11:35 PM
To: tgeorgiev at is-bg.net; asterisk-ss7 at lists.digium.com
Subject: Re: [asterisk-ss7] chan_ss7

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 at is-bg.net> 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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