Just as a remark, there were a lot of BRI lines in USA using 64k. We ran ISDN videoconference stations in AT&T CALA to US in early 2000's (using AT&T and Sprint networks). They'd some DS1 with 64k channels to support international Nx64k calls with ANSI ISUP, but I'm not sure if that network remains active. Anyway, that was just a comment about this interesting subject. Gustavo On Dec 9, 2011, at 11:36 AM, Jan Berger wrote: > 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 -------------- next part -------------- An HTML attachment was scrubbed... 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