[asterisk-ss7] SS7 + T1 on Asterisk?

[Jan Berger]

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 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20111209/4d861258/attachment.htm>