[asterisk-ss7] chan_ss7

Teodor Georgiev tgeorgiev at is-bg.net
Wed Mar 1 01:51:07 MST 2006



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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