whew - i was choking on "SS7 on BRI" typical in US/Canada is T1 or DS0A (inside the switch office only); some other 56/64 interfaces are DSCS and V.35 http://en.wikipedia.org/wiki/DS0A Asterisk users need only be concerned with T1, however On Fri, Dec 9, 2011 at 12:43 PM, Jan Berger <janvb at live.com> wrote: > Sorry - correcting myself - 56kbps is used on T1's with CAS. > > ________________________________ > From: janvb at live.com > To: asterisk-ss7 at lists.digium.com > Date: Fri, 9 Dec 2011 18:36:33 +0100 > > Subject: Re: [asterisk-ss7] SS7 + T1 on Asterisk? > > http://en.wikipedia.org/wiki/Digital_Signal_1 > > I believe it's BRI lines that uses 56kbps and your right that SS7 on BRI > have some usage in US. > > Jan >> Date: Wed, 7 Dec 2011 20:12:22 -0200 >> From: marcelo at m2j.com.br >> To: asterisk-ss7 at lists.digium.com >> Subject: Re: [asterisk-ss7] SS7 + T1 on Asterisk? >> >> Typically T1 (american) signaling ss7 links run at 56kbps instead of >> 64kbps. >> If your switch can run 64kbps links over a T1 timeslot, than the only >> remaining variable is ITU versus ANSI ISUP. They are incompatible >> (different message formats due to different network address sizes and >> other details). >> We use ITU ISUP all over the place without trouble. If the switch can do >> 64kbps links and ITU ISUP, then you should be able to use all existing >> E1 direct connection samples (without STP), except for the obvious E1=31 >> timeslots while T1=24 timeslots difference.. >> ANSI might work. I won't go there because I have zero experience with >> ANSI SS7/ISUP (stability wise). >> With 2 T1 and a single signaling link it should allow for 47 voice >> channels and one signaling link. >> >> Search for libss7 ansi 56kbps for the most difficult scenario. But if >> you can do ITU ISUP + 64kbps links, I would suggest that instead. >> We hardly see people talking about ANSI ISUP setups on this list, so it >> could be far less stable (at least it seems to get less usage). >> >> On 12/07/11 16:25, Matt wrote: >> > In this case, our supplier is ourselves. We have a DMS100, but the >> > switch guy is someone other than myself - I am the IP guy. >> > >> > So basically if I understand you properly, I should be able to do the >> > SS7+T1 and get proper operation, provided the configuration on both >> > sides is right. >> > >> > On Wed, Dec 7, 2011 at 1:06 PM, Marcelo Pacheco<marcelo at m2j.com.br> >> > wrote: >> >> If the DMS100 switch can talk signalling directly with Asterisk, >> >> without an >> >> STP, it should be possible to use a single timeslot for ss7 signalling, >> >> so >> >> with 2 T1 you could have 47 voice calls and one signalling channel. >> >> This is >> >> common with E1 setups. Also with E1 its common for a timeslot to be >> >> statically switched over to an STP (semi permanent call), allowing for >> >> access to the signaling network without a dedicated physically separate >> >> signaling link, but that's not usual in T1 land. >> >> >> >> But what you ask is technically possible... However its important to >> >> PROPERLY LEARN SS7 terms to be able to communicate with your supplier. >> >> SS7 is a CARRIER LEVEL PROTOCOL. However people insist on winging it >> >> without >> >> proper training. >> >> Its like trying to become a backbone internet provider without properly >> >> learning inter and intra network routing protocols (like BGP and OSPF). >> >> >> >> If you knew the general SS7/ISUP knowledge, you could quickly find the >> >> information you're looking for on Google. >> >> >> >> PS: I live in E1 land... I'm just quoting information from the top of >> >> my >> >> head. I have no need for T1+SS7. E1+SS7 is a little simpler with >> >> Asterisk >> >> than T1+SS7 due to 56kbps data links, ANSI ISUP/SS7 and some other >> >> quirks. >> >> >> >> Good luck. You'll need it. >> >> >> >> >> >> On 12/07/11 14:47, Matt wrote: >> >>> If I were to get a 2 span T1 card for Asterisk... and connect it to a >> >>> Nortel DMS100... can I run call traffic over the T1 and run SS7 >> >>> signaling FOR the T1 over the other port? >> >>> >> >>> Is there documentation on doing this anywhere? >> >>> >> >>> -- >> >>> _____________________________________________________________________ >> >>> -- 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 >> >>> >> >> >> >> -- >> >> _____________________________________________________________________ >> >> -- 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 >> > -- >> > _____________________________________________________________________ >> > -- 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 >> > >> >> >> -- >> _____________________________________________________________________ >> -- 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 > > -- _____________________________________________________________________ -- > 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 > > -- > _____________________________________________________________________ > -- 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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