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Legend PRI to Cisco Voice T1 setup 1

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bgflyer

IS-IT--Management
Jul 27, 2007
5
US
I need to figure out how to connect a Legend system to a Cisco 2821 router in order to use VOIP between offices.

My T1s on the Legend show up as:
A Slot Type Format Supp Signal LineComp
A 5 T1 D4 ZCS Rob_Bit 1
A 11 PRI ESF B8ZS DMI-MOS 1
A 16 PRI ESF B8ZS DMI-MOS 1

I want to have the 2821 take the place of the PRI T1 between offices, however I cannot make the 2821 talk to the PBX even over the existing PRI line. Any thoughts on how to make either CISCO speak PRI and connect, or how to make the Legend communicate using a standard that I can match to the 2821?

Cisco does not know what DMI-MOS is...anyone have any resources with more info on what this means?
 
Is the "CISCO" it's own switch?

The Cisco must be able to emulate one of the FEW things a LEGEND can interface with, AND NOT BE A NATIONAL PRI.

If you look at SWITCH TYPE in PRI, you will see the choices.

Doesn't CISCO stand for "can i still call out"?
 
Good question...I don't know what a Legend can interface with that a Cisco 2821 will emulate. The ISDN options for Cisco show:

primary-4ess Lucent 4ESS switch type for the U.S.
primary-5ess Lucent 5ESS switch type for the U.S.
primary-dms100 Northern Telecom DMS-100 switch type for the U.S.
primary-dpnss DPNSS switch type for Europe
primary-net5 NET5 switch type for UK, Europe, Asia and Australia
primary-ni National ISDN Switch type for the U.S.
primary-ntt NTT switch type for Japan
primary-qsig QSIG switch type
primary-ts014 TS014 switch type for Australia (obsolete)

The options on the legend show:
4ess
5ess
dms-250
dms-100
dex600e
merlin-ntwk
merlin-pbx

Currently I have two merlin pbx's connected using merlin-pbx. However, I want to put the router inbetween. However attempts to use 4ess do not allow the merlin to talk to router.

Also a side question...all of our ds1 ports on the merlin go to an external csu-dsu. Is this common and necessary?
 
primary-4ess or 5ess may work from the Legend to the Router.

Doing it that way however, would probably kill your UDP.
 
DMI-MOS stands for "Digital Multiplexed Interface-Message Oriented Signalling".

ISDN and X.25 messages use either a "message oriented" (character oriented) or bit-oriented signalling (known as BOS).

The DMI piece is a proprietary protocol that AT&T has been using since time out of mind. It's used with Definity DCS as well as Legend/Magix networking.

Bottom line? The two PRI entries shown that use the DMI-MOS signalling are endpoints to other networked Legend or Magix. If you want to supplant the PRI circuit (which is actually a straight T1...not a true PRI) with a router that uses an IP VPN, you'll have to program the Cisco for "T1 circuit emulation". I've got a Cisco link for that if you need it, but you should be able to find it easily. Essentially, the Cisco will be used to create a tranparent pipe between DSX interfaces end-to-end.

Our PBX defines the circuit as a PRI, but it's really a bunch of B-Channels with a proprietary protocol running on the 24th channel. It's because of the proprietary protocol that we have to use straight T1s to carry the signalling properly end-to-end.

Trying to set this up as a true PRI with 5ESS (or whatever) will not work.

Write back and I'll help any way I can.

Tim Alberstein
 
Thanks Tim!

I learned something new today. I did not realize the Network Signalling Protocol for networking Legend/Magix & Definity used a special "PRI". That explains a few curiousities about networking with non AT&T/Lucent/Avaya PBXes.

So no one else uses the DMI-MOS?

I can see where the Cisco would go nuts trying to do real PRI to a special "PRI", and this would probably hold true for other DS1 router/DACS/MUX equivalents, where one is doing a similar configuration, networking these switches.

----

bgflyer,

When you connect the 100D module to the Cisco, you don't need the CSU, unless it was doing something else besides being a CSU. What I mean by that is, some CSUs allow parameter differences between DS1 facilities. An example would be the incoming DS1 coming from the provider was configured as D4/SF, B8ZS and you needed D4/SF, AMI, because your equipment did not support B8ZS line code. Using the CSU in this way, it fixes the "problem". So unless you have already checked the parameters of the DS1, it never hurts to verify this stuff!

....JIM....
 
It sounds like my plan will not be possible then, I was hoping to overcome the 3 T1 limitation on our Legend system to add in a newer voip system and allow interoffice dialing and PSTN access to the old system while implementing the new.

Does anyone know how to make a Cisco router appear like a PSTN connection? This is the other option for integrating the system, keeping the Merlin system separate, and moving the PSTN T1 to the Cisco router to add the VOIP system into the mix.
 
I'm not sure how you were expecting to overcome the three T1 limition of the Legend/Magix. It's not the protocol that's killing you, it's the total number of DS1 boards that are allowed in the switch. Whether these boards talk to a T1, a PRI or a Cisco is immaterial.

I need to know what Cisco box you're using before I could assist with a config. There's the Unity Gateway, the IP PBX platform (usually associated with CallManager) and of course, their garden-variety large routers running MultiService OS doing simple voice switching.

You realize that PRI or POTS is NOT the only way to integrate all switches. It would be a lot of work, but you could change the PRI programming to straight T1 and use good old E&M Wink between PBXs. For the Legend/Magix users, people would have to touch or dial a "tie line" code, then the remote extension. The Cisco could then be just another ordinary T1 E&M Wink endpoint with it's own dialplan.

It might go like this: I'm at Legend Site A and want to talk to Legend Site B. I dial a tie line code of "810". When I hear dialtone, I dial "2", then the remote extension number. If I want to reach someone at Legend site C, I dial "810", then "3", then the remote extension. To reach a VoIP extension at the fourth side, I dial "810", "4", then the VoIP station number.

The trunk access code could be reduced to a single digit, making the strings appear more like "82", "83" and "84". In any case, it would take some getting used to. But it sure beats the pants out of dialing some seven digit number to reach the other site. And if the Cisco sits in the middle of all three PBXs, it allows for a full integration...your primary goal. The weird dialing scheme would disappear as soon as you were ready to cut completely over to Cisco.

Tim Alberstein
 
Thank you Tim!

I am using a Cisco 2821. The eventual configuration would be linked to CallManager. The problem is, and the reason to overcome the 3 T1 limitation is that I have 2 remote sites using the central voice mail, and 1 T1 going to the PSTN. The problem is I want to add a new VOIP system for a new office and use the PSTN T1 I currently have for both.

My solution: remove the PSTN from the central PBX and place it on the 2821. Then connect the central PBX to the 2821 to route outgoing calls through the PSTN (or to the remote office via VOIP).

My fear: We have enough funds to implement the new remote site. However, the Cisco vendor does not have Avaya experience and is leaving the Avaya programming to us to either figure out ourselves or hire out. Since we are going to implement the VOIP system at the remaining 3 offices within the near future (months), I hate to spend a lot of money on programming the avaya system, so I want to do it myself. Otherwise I would just wait until we have the funding to do all 4 sites at one time. But even that is not preferable because of the downtime in getting 4 new sites up and running.

Since I could take down one of the three offices on Fridays, I thought of doing an alternative and having the Cisco sit off the Legend system and have calls route out through it. However, this seems more problematic for many reasons.

Bottom Line: Just for testing to see and calm my nerves that yes, the Legend can see a Cisco router and think of it than nothing other than the PSTN and get calls to route out using just basic call routing functions on the router (leaving the eventual Call Manager setup to the experts)will ease my fears of spending 50,000 and then having to wait 3 months anyway to add the other sites onto the callmanager system in order to bring our new site online.
 
PS. I work for a non-profit agency...thus the high concerns with spending money that often just isn't there.
 
So you're a non-profit eh? Well, it looks like you've come to the right place for help.

OK. So are you saying that you want to replace the D4/ZCS "straight T1" circuit with a Cisco emulation of a CO-provided T1? That's pretty straightforward.

I assume you've got a T1 WIC in that 2821, right? Go to Cisco's website and do a seach for T1 CAS with E&M Wink Start signaling (preferred over loop-start emulation for reliability). We're gonna want to present the DCE or CO side of the circuit to the Legend/Magix. You should be able to find a script without too much hassle. But let me know if I'm full of it...

After that, it's a simple matter of figuring out whether you need a straight or crossover T1 cable, and who (the Cisco or the Legend) will provide the clock source over this "fake" T1.

Sorry about the delayed response. Busy day.

Tim Alberstein
 
Do I understand correctly that we *cannot* use a PRI between a Merlin Legend and a Cisco 2821 voice gateway, and that it must be a T1-CAS ?

Just wanted to clarify...thanks!
 
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