Sorry - correcting myself - 56kbps is used on T1's with CAS. From: janvb@xxxxxxxx To: asterisk-ss7 at lists.digium.com Date: Fri, 9 Dec 2011 18:36:33 +0100 Subject: Re: 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 -------------- next part -------------- An HTML attachment was scrubbed... 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