[asterisk-ss7] chan_ss7

[leonimar cape]

Hi,

Is there any updates on the version of the chan_ss7? I
am just really excited to see the version with the SUS
and RES isup messages being supported.

Cheers!

Leonimar Cape

--- Tim Danner <tdanner at pacwest.com> wrote:

> SUS means suspend, till a resume (RES).  They both
> originate from a
> line.  This is normally by flash calls, instead of
> hang up.
> 
> Suspend the entire trunk path, to allow a third
> party call or place on
> hold (to be brought on line).
> 
> Resume the call once established.
> 
> 
> -----Original Message-----
> From: leonimar cape [mailto:leo_mac_ph at yahoo.com] 
> Sent: Tuesday, February 28, 2006 5:18 PM
> To: asterisk-ss7 at lists.digium.com
> Subject: [asterisk-ss7] chan_ss7
> 
> 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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