voice problems with cisco and norstar 1

[schrunk]

Hello,

My name is Brian and my situation is this:

My company has two locations and recently switched over to cisco 2620 routers on both ends. We have a Norstar MICS ver. 4.1 as our voice system and a point to point T1 as our circut between locations. Our routers contain VWIC 2MFT-T1-DI cards at both ends. I've had no luck running voice over this circut since brining in the new routers. I'm using e&m wink (clearchannel) signaling with B8ZS line coding and ESF framing. I've turned off the dsu on the Norstar on both ends. I'm trying to use channels 21-24 on the T1. When I try to pick up a line, I receive the message "Line in use" from the Norstar. Any ideas would be much appriciated!

My cisco config is listed below:

version 12.0
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname PARRTR
!
ip subnet-zero
!
isdn voice-call-failure 0
!
!
controller T1 0/0
framing esf
clock source internal
linecode b8zs
channel-group 0 timeslots 1-20
description PPP T-1 to Lynwood
!
controller T1 0/1
framing esf
clock source internal
linecode b8zs
tdm-group 0 timeslots 21-24 type e&m
description Voice to Lynwood
!
!
interface FastEthernet0/0
ip address {ip addr} {ip addr}
no ip directed-broadcast
speed auto
half-duplex
!
interface Serial0/0:0
description Point to Point Lynwood
ip address {ip addr} {ip addr}
no ip directed-broadcast
encapsulation ppp
!
interface FastEthernet0/1
no ip address
ip helper-address {ip addr}
no ip directed-broadcast
shutdown
duplex auto
speed auto
!
interface Serial0/2
description Fontana Point-to-Point
ip address {ip addr} {ip addr}
no ip directed-broadcast
encapsulation ppp
service-module t1 clock source internal
service-module t1 timeslots 1-24
!
ip classless
ip route {ip addr} {ip addr} {ip addr}
ip route {ip addr} {ip addr} Serial0/0:0
ip route {ip addr} {ip addr} Serial0/2
no ip http server
!
!
line con 0
password root
login
transport input none
line aux 0
line vty 0 4
password root
login
!
no scheduler allocate
end

 

[jerryreeve]

I cannot talk to your CISCO side of the problem but may be able to give you a hand on the NORSTAR side. My first suggestion is to ask the same question on the NORSTAR forum since my time in the office sporatic and there is usually someone there who can give you a suggestion.
Second in regards to the configuration of the norstar, how were you planning to use this? by that I mean did you intened to use the access code for the pool to be the first digit dialed to be followed by the rest of the digits for the station at the other end of the t1? or had you planned a consolidated numbering plan where every telephone you have would have a different extension with the stations located at site a to start with a 2 and the stations located at site b to start with a 3?
How is your systems presently set up for your t1 trunks 21-24 specifically under Lines->(021-024)->Trk/Line Data->Trunk type, line type, trunk mode, answer mode, and answer with DISA? (they should all be same)
Also for the target lines Lines->(145-272)->Line type,Rec'd #,If busy,and Primeset? (just the first one you are using to test with and get you started)
and under System programming->rec'd # length? lastly under Hardware->show module->ksu->kards on ICS->card(1 or 2)->t1, framing, internal csu, line coding, clocksrc? I know a lot of questions to stat off with JerryReeve
Communications Systems Int'l
com-sys.com

 

[trusnock]

We have an almost identical scenerio, with the same symptoms. Does anybody know what the outcome of this thread was? I know this is a longshot, since the thread is three years old, but I'm having trouble finding any support on this setup.

We are using two Cisco 1721 routers with VWIC-2MFT-T1-DI cards configured to mux channels 20-24 through to our Norstars. The data channels (1-19) are working fine, but the Norstar reports "Line in use" when we try to grab a line. I'm the data guy, so I don't have many details on the Norstar setup, but I do know that it works fine when we remove the routers from the picture and let the Norstars see the whole T1. Everything is ESF, B8ZS, E&M Wink; we tried several different clocking scenerios.

Any help you can give will be greatly appreciated!

Thanks,
-Tom Rusnock

 

[rtfmdude]

Tom,

Well, the cisco config at the top is missing a significant portion of the tdm-group config, you need to specify the channels that you want to pass through on both controllers, not just one. Post your config and let's have a look.

Eric

 

[trusnock]

Here is the config from one of the Cisco 1721 routers. The other side is identical, except the clock source is line on controller T1 0 (I also tried it with the clock coming from the main Norstar, and everything recovering clock accordingly). The Norstar can see a loop at the far end when we install a loopback plug in place of the far-end Norstar, so I am fairly confident that it is MUXed correctly in the Ciscos.

The Norstars are running version 6.1. We are using a 6-port combo card, and it is plugged into slot 1 on the main cabinet.

Thanks for looking at this for me...

!
!controller T1 0
framing esf
clock source internal
cablelength short 133ft
linecode b8zs
channel-group 0 timeslots 1-19 speed 64
tdm-group 1 timeslots 20-24
description T1 from Telco
!
controller T1 1
framing esf
clock source internal
cablelength short 133ft
linecode b8zs
tdm-group 0 timeslots 20-24
description T1 to PBX
!
!
interface FastEthernet0
ip address 10.0.0.3 255.255.255.0
speed auto
full-duplex
!
interface Serial0:0
description Fractional T1
ip address 10.0.99.1 255.255.255.252
!
ip classless
ip route 0.0.0.0 0.0.0.0 10.0.0.1
ip route 10.0.1.0 255.255.255.0 Serial0:0
!
no ip http server
!
snmp-server community 4halac-snmp! RO
snmp-server enable traps tty
connect PBX T1 0 1 T1 1 0
!
!
!
control-plane
!

 

[rtfmdude]

Tom,

Here's a cisco doc on this:

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a008010f05d.shtml

I actually supported this stuff for awhile, and frankly, didn't understand 1 thing - if we're muxing/demuxing at the bit level, why does cisco have another option when you're passing e&m control data across the t1? Dunno, doesn't seem to make sense to me. Anyway, try this on both controllers:

controller t1 x/x
tdm-group x timeslots 20-24 type e&m

 

[trusnock]

Hmmm... That helps, but I'm not sure it applies to the 1721. The "type e&m" option is only available when the controller is in "mode cas" (Channel Associated Signaling). However, there is no "mode cas" command available under the controller on my 1721. We're running a very recent release, 12.3(11)T3, but I'm digging now to see if it appears in some other IOS version.

Also, I thought we could circumvent the whole robbed-bit problem by using PRI mode (yes, strange, a fractional PRI, but I've seen it work in other secenerios as long as channel 24 is included). Maybe I'm misunderstanding robbed-bits-- I thought they weren't used at all in a PRI-- but we tried PRI and it didn't work either.

 

[trusnock]

Well, after more than a week working with Cisco TAC, they are now telling me that the 1721 router does NOT support "voice" TDM, only "data" TDM. (that seems pretty pointless... Have you ever found yourself wishing you could share a T1 between two fractional T1 DATA circuits?!)

They say we'll need to use 1751 routers (Voice over IP routers) to handle the Channel Associated Signaling (CAS), even though what we're doing has nothing to do with VoIP.

If anyone has any information to the contrary, I'd love to hear it! In the meantime, we're verifying the minimum 1751 configuration that will accomplish this, and planning to give it a go with those routers.

 

[rtfmdude]

wow. never knew that.

"Data only support for the Cisco 1721"

http://cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800f9d37.shtml#other

what a piece of garbage!

 

[trusnock]

Yes, and there are only one or two vague references on Cisco's entire site to indicate that the 1721 won't do "voice TDM". When I originally read that, I thought they just meant it couldn't terminate a voice call because it's not a VoIP router.

They are so careful to explain that the Drop-and-Insert function is strictly handled in the VWIC card, and never leaves the WIC slot. It even has a high-availability mode where it will maintain the multiplexing through a reload of the router! All this lead me to believe (firmly) that they were talking about a lack of VoIP capability, and not a lack of CAS capability.

I've been disappointed before (many, many times) by Cisco's lack of clarity of documentation, but this is a bad one. I really wish they would make it more clear, so their Presales people don't continue misleading vendors, and so vendors have some hope of discovering this limitation BEFORE showing up at a customer site with the wrong equipment in hand.

End of rant.

Thanks again for your help with this issue. I'll let you know how it turns out (whether we go with a 1751 router or just an external MUX).

-Tom Rusnock

 

[SYQUEST]

You won't have this type of problem with ADTRAN. They know how to do T1 right! They don't play games like Cisco and then want to charge you for it!!

....JIM....

 

[trusnock]

I'm all ears. I've been working with Cisco products for 9 years now, and I've put up with the same bull from them the whole time. What do you recommend from the Adtran line?
-Tom

 

[SYQUEST]

I would give ADTRAN a call to find out what would best fit your application. The only number I have at hand is customer care, 888-423-8726, but that at least gets you in the door. Let us know what they recommend.

Hope this helps!

....JIM....

 

[computerhighguy]

I don't get it. Why don't you just turn the second controller on the VWIC into a PRI on both ends and let the router use VoIP to complete the calls? It would take about 10 minutes to setup and you would get more 4 channels of voice and could use all the channels for data?


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)

 

[ISDNman]

Cisco's a data company. Even some of the definitions on their website related to Telephony are wrong.

I am not syaing they are "bad" but it is no surprise they couldn't get the Robbed bit to work and then pulled the feature (that is my guess as to what happened).

I agree that Adtran has a clue and has good support.

 

[trusnock]

Computerhighguy,
The 1721 routers that we were attempting this with are not VoIP-capable. Since the customer's needs are pretty simple, we hoped to do this with 1721 routers since they are available for 1/2 to 1/3 the cost of Cisco's least expensive VoIP router (the 1751-V).

I think your suggestion would be feasable with the 1751-V, but I think it only comes with enough DSPs for 2 channels, so even that router would need to be upgraded to accomplish the goal.
-Tom

 

[computerhighguy]

trusnock,
Get a couple of MC3810s off of ebay. They are extremely cheap and will do what you want at a reasonable price as long as you don't use more than 24 calls at any 1 time. The CLI is a bit slow, but who cares, the units are only a couple of hundred dollars. I use them for lab equipment and I set up our customers who are looking for an inexpensive way to get into VoIP. I usually just purchase an extra unit, rather than try to bring hem under contract.


It is what it is!!
__________________________________
A+, Net+, I-Net+, Certified Web Master, MCP, MCSA, MCSE, CCNA, CCDA, and few others (I got bored one day)