Florian, it looks like you follow this list not for a long time. chan_ss7 devs are also subscribed to it. While this question is asterisk ss7 specific, there is a reason to ask here. If someone experienced the given, he may respond. 2011/9/2 <florian at gruendler.net>: > Wasim and general audience, > > > > I felt addressed in your post and I react before I get bashed and a policy > discussion starts over my statement. I give in to the point that this list > is probably the only good forum at the moment for Asterisk and SS7 related > topics and all flavors of software should be welcome. But I stick to the > point that I suggested Raul to go to ?his? development sponsor with the > specific question. > > > > In an open source development model the author of code wants it to be public > and there is no stealing of ideas. Open source is about providing the source > of software for public use, review and enhancement, but not about having the > right for anyone to get support for no money and take a free ride to realize > low cost commercial offers ?just because you can?. The total cost of > ownership of open source software mainly consists of external support > efforts or at least of having knowledgeable people on the payroll. Privatize > profits with offering based on free stuff and socialize cost for support is > absolutely not on the agenda of open source software because it would mean > that people working in the industry have an empty fridge. > > > > Rauls question was VERY specific on a chan_ss7 parameter and its internal > crunching logic. He should get paid vendor support or trust on the vendor > goodwill to dedicate time of subsidized people to the list, which Digium > COULD do here from hardware sales profits but other software-only shops are > a bit out of luck. I haven?t really seen any Digium staff giving replies > here so apparently the don?t want to go down that road to push ?SS7 at > home?. Digium is probably commercially better off to dedicate their SS7 > resources to paying OEM customers like Sysmaster. > > > > Are we now all friends again? > > > > So long, Florian > > > > > > Von: asterisk-ss7-bounces at lists.digium.com > [mailto:asterisk-ss7-bounces at lists.digium.com] Im Auftrag von Wasim Baig > Gesendet: Freitag, 2. September 2011 14:52 > > An: asterisk-ss7 at lists.digium.com > Betreff: Re: [asterisk-ss7] Nature of Address chan_SS7 > > > > AFAIK, asterisk-ss7 has had discussions on all SS7 options for asterisk > including libISUP, lib_ss7, chan_ss7, ss7box, SMG etc ... > > > > Has the list policy changed to be solely lib_ss7? > > > > -wasim > > On Fri, Sep 2, 2011 at 14:36, <florian at gruendler.net> wrote: > > Raul, > > > > You are on the wrong channel with this question. You question is not > Asterisk related, much less lib_ss7 related. You ought to ask to the people > behind chan_ss7. It may be okay to discuss a general mobility topic on a BMW > list, but you aren?t getting much cheers if you discuss your Mercedes > problem on the BMW list despite the fact that both are cars. > > > > Regards, Florian > > > > PS: If you are a telco and you have local customers being switched to > another local customer (=physically connected to the same PBX), you > automatically have a dedicated exchange for that and a separate > trunk/linkset for non-local calls being sent off to your national or > international exchange. Nobody would want what you want anyway so what are > you trying to do here? Is this just an academic lab question or why does > your network planning require this? > > > > > > > > Von: asterisk-ss7-bounces at lists.digium.com > [mailto:asterisk-ss7-bounces at lists.digium.com] Im Auftrag von Raul Baldeon > Gesendet: Donnerstag, 1. September 2011 23:42 > An: asterisk-ss7 at lists.digium.com > Betreff: [asterisk-ss7] Nature of Address chan_SS7 > > > > Hello list, > > I have an issue with chan_ss7. My current configuration only support > national calls or local calls; (I can switch changing the parameter noa from > linkset). > Is there a way to work with both of them? > > Here you are my configuration: > > [linkset-siuc] > enabled => yes > enable_st => no > ;use_connect => yes > use_connect => no; When use_connect = yes; There is no ringback in SS7 -> > SIP > hunting_policy => even_mru > context => ss7 > language => da > t35 => 15000,timeout > subservice => auto > variant => ITU > noa => 0x1; (0x1 => for local calls and 0x2 for national calls) > > [link-l1] > linkset => siuc > channels => 1-15,17-31 > schannel => 16 > firstcic => 1 > enabled => yes > sltm => no > > > [host-mihost] > enabled => yes > opc => 0x391 > dpc => siuc:012 > links => l1:1???????????????????????? ;span 1 of dahdi/system.conf > globaltitle => 0x00, 0x00, 0x01, 114509090 > ssn => 7 > > Best Regards > > Ra?l > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > -- > wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | > peace be upon you ... > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-ss7 >
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