FXO and T1 ports on a router

Sludgebucket

I'm having to use an FXS, FXO, and T1 ports on this voice router. I see the FXS and T1 ports, as they'rel clearly labeled. But I'm not sure which the FXO port is, and I'm told to "connect an RJ-11 phone cable fromt he lowest-numbered FXO port o the router to a port on the Adtran FXS card. I need a loose phone cable to connect up with? Now I'm confused.

I have empty regular fast ethernet slots, empty WIC slots (which are the T1 slots), and that's it. Where would I see FXO ports? I'm checking connectivity between this router and a PSTN, which for this lab is done with an Adtran.

dtlokee

Just like on the FXS VWIC the FXO VWIC is clearly labeled next to the FXO interfaces. They will be RJ-11 interfaces.

mikej412

Do a show voice port summary if you need a hint as to which slot to look at.

You could also have a VIC-2DID -- 2-port Direct Inward Dial (DID) voice/fax interface card -- or the VIC2-2FXO, VIC2-FXO, EM-HDA-4FXO, EM-HDA-6FXO, EM-HDA-3FXS/4FXO etc., depending on your router model.

Check out show voice port summary Command: Examples and Voice Hardware Compatibility Matrix (Cisco 17/26/28/36/37/38xx, VG200, Catalyst 4500/4000, Catalyst 6xxx) for more information.

You can also search the Cisco web site for your specific router model if you're not familiar with the numbering of the modular router slots/modules.

Sludgebucket

A lot of my problems, I just discovered, was that I had an Adtran only set up for ISDN, not T1.

Sludgebucket

Perhaps someone could clarify some dark areas for me:

If you have a voice-enabled router, and to it are connected one IP phone and one analog phone, you'll need a pots dial peer for each phone, right? The router is connected to an Adtran, and on the other side of it is another pod, identical in every way. So - if I'm right - you'll have your existing two pots dial peers for each side, as well as one (?) voip dial peer to connect across the PSTN/Adtran? So for each router you'll have three dial peers, or more than this?

I'm trying to come up with a really easy scenario which will clear up some things in my mind before I go any further. Thanks.

dtlokee

You will need a pots dial peer for the directly connected analog phones. I am assuming you are creating an analog trunk to the adtran which means you will need a pots dial peer here as well pointing to the ds0 group you created (if the T-1 controller is in slot 1 interface 0 and you made ds0-group 1 it would be 1/0:1)

so

controller T1 1/0
linecode b8zs
framing esf
ds0-group 1 timeslots 1-24


dial-peer 1 pots
destination-pattern 1111
port 1/0:1
no digit-strip

that will forward the call for the dialed string of 1111 across the T1, you will need a local dial-peer for 1111 on the remote side that points to the voice port on the local router (fxs). This is also assuming you are using E&M signaling on the T1, if not specify on the ds0-group.

Sludgebucket

I follow you on most of that. I haven't done much with the "digit-strip" yet, but that's next.

Here is what I have for this particular lab topology, but I need to know if it's sufficient to make it work. (Btw, how can I draw a topology out in order to display it for a diagram in this forum?)

2 pods
1 Adtran (my PSTN)
Each pod consists of: 1 Voice enabled 2800 router, to which is connected one IP phone and one
analog phone

The dial plan for my first pod is:
Ext. #s - 5030 to 5059
First E.164 DID Number - 510555030
Voice Mail Ext. # - 5555058

The dial plan for my second pod is:
Ext. #s - 5060 to 5089
First E.164 DID Number - 5105555060
Voice Mail Ext. # - 5555088

---

So, for example, my first pod has three dial peers (I'm still at the elementary stages on some of this):

dial-peer voice 1 pots <---[this would be for the analog phone, I presume]
destination-pattern 5555058
port 0/2/0

dial-peer voice 2 pots <---[this would be for the IP phone, I presume]
destination-pattern 555[4,5,6]...
port 0/0/0:23
forward-digits all

dial-peer voice 3 pots <---[this would be for incoming calls, I presume]
incoming called-number 5555...
direct-inward dial
port 0.0.0:23

Does this look like it would work? And when I try to hook up a T1 line and get the phones to talk across the PSTN to each other, what other dial-peer should be implemented?

Lastly, when you're configuring a voip dial peer, you create something like:
session target ipv4:10.19.0.1

Where do you get the IP address? I know it's of the other router, but if the lab you're doing doesn't just give you the router IP, how would you go about getting it if you have to configure it in the first place?

Boy, I wish I had a tutor for this stuff. I'd gladly pay for one.