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@xxxxxxxxx> 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@xxxxxxxxx] > > Sent: Wednesday, March 01, 2006 12:24 AM > > To: asterisk-ss7@xxxxxxxxxxxxxxxx > > 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 > > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > protection around > > > http://mail.yahoo.com > > > _______________________________________________ > > > --Bandwidth and Colocation provided by > Easynews.com -- > > > > > > asterisk-ss7 mailing list > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > _______________________________________________ > > --Bandwidth and Colocation provided by > Easynews.com -- > > > > asterisk-ss7 mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com > -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
|