[asterisk-ss7] chan_ss7

[Tim Danner]

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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