Basic questions for CCNA VOICE

neurotic

I'm new w/ VOICE and there are things that i don't quiet understand.
Say for example in cisco 2900 series, there are voice/wan interface that we can add like VIC3-4FXS/DID.
If i'm not mistaken, FXS port is intended to connect to phones. These are analog phones, correct?
On the other hand, FXO ports are intended to connect to PBX system.
So if i have a cisco 2900 series w/ a VIC2-4FXO module, is that the port where i connect the cable from the provider going to my branch office that also has VIC2-4FXO module.

Also, what is DID? it says it's a service provided by telephone company that enables callers to dial directly into an extension on a PBX.
so does that mean the Cisco 2900 series router is our PBX?
So if i have a T1 line from my internet service provider, is that the same line that we will be using to connect to the ISR gateway?
Also that is a PRI, BRI?


Apologies w/ these stupid questions.

Thanks!

davenull

Disclaimer: I'm no expert by any means and I still have a limited real-world experience but I'll try to explain a few things.

About FXS/FXO:

Think of FXS/FXO ports as male and female. If one end has an FXS, then the other end has to have an FXO port. FXS provides a dial-tone, FXO expects to receive a dial-tone. Yes, you connect analog phones to an FXS port on the router, you can think of your analog phone having an FXO port built-in.

A PBX functions as a somewhat of a scaled down private PSTN, it provides dial-tone, therefore you can use an FXO port on your router to connect to it.

You can also use an FXO port on your router to connect to the PSTN. Again, you can think of PSTN having an FXS port on their side that provides you with a dial-tone. You are limited to one call per port with FXS/FXO.

About DID:
First of all, DID is only applicable to POTS dial-peers. There's no need for DID with VoIP. Mind you, everything we talk here is POTS, including PRI - just because your call is digitized doesn't make it a VoIP call.

DID is needed if your company is big enough that you want people from outside to call a particular phone on the inside. To understand DID it helps to understand the limitations of an FXO port first. Let's say you have a router with an FXO port connected to the PSTN. As funny as it sounds that router (just like an analog phone) does not know it's own phone number - an FXO port doesn't receive called number (DNIS). An FXO port on a router acts just like an analog phone - it receives an electrical signal from PSTN causing it to ring, but PSTN can't provide it with called digits, so the router doesn't really know where to send that call. For this reason, you'd usually configure something called connection plar that automatically rings one phone (e.g. receptionist's), or all phones configured with the same shared line (viable for a small company), or you can have connection plar point to a hunt pilot. The router's logic here goes like this: if my FXO port lights up, always ring this preconfigured number; whereas what we want is this: if my port lights up and the called number is x, ring that called number (possibly after manipulating it).

If you really want DID functionality, but can't justify purchasing PRI, you can talk to your phone company and install an FXS/DID card in your router along with FXO. Both of these cards will connect to PSTN. This was what confused me when I was studying this - what, since when are we using an FXS card to connect to PSTN? The reason it works is that you will use FXO card for outbound calls to the PSTN, and FXS/DID for inbound calls. Unlike FXO, FXS/DID gets called number because, again, when you dial from an analog phone, the FXS port on the router obviously gets dialed digits.

Unlike FXO port, PRI is able to send both called and calling number, so the router has no issues routing the call to the inside; however, you have to configure direct-inward-dial under an incoming pots dial-peer to make the router to immediately try to match an outgoing dial-peer instead of providing dial-tone to the person calling from PSTN. You can research this more by looking up two-stage vs one-stage dialing.

About T1:
You can configure a vwic (voice/wan interface card) T1 to be:
1. A plain WAN data interface using channel-group command. There are no voice channels involved here.
2. A digital voice interface using ds0-group. This is going to use something called CAS. You get all 24 voice ports but it has less features than PRI ISDN.
3. A digital voice interface using pri-group. This is what you usually see companies use to connect to the PSTN. PRI uses CCS with 1 channel reserved for signaling.

A BRI is just like PRI but has only 2 voice channels (B-channels) and 1 signaling channel (D-channel), I don't think it's used much.

jeremus

Are the DID Trunks from a telephone service provider similar to the trunks of PRI-T1, say 24 channles ? Are these 23 b-channels and 1 d-channel for a PRI-T1 are the same as trunks of the DID service?
Dou you still need a DID service offered by telephone companies that enables callers to dial directly an extension on a PBX or packet voice system if you have already a PRI-T1 service?
Do they have the same function between DID Trunks and Voice Channels of PRI T1?
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davenull

Well, you dug out my year old post. The definition of DID I put there is not how I would describe it now.

When people talk about DID they talk about numbers, such as:

DID number - an internal number that can be dialed directly from outside (an analogy would be a public IP address you buy from your provider).

Non-DID number - an internal number that can't be dialed directly from outside, it can be reached only via auto-attendant IVR or receptionist (an analogy would be using NAT).

You can buy numbers from your telephone service provider just like you would buy public IPs. If you bought a range of numbers then the service provider is going to send those calls your way. Now it's up to you to route those numbers to endpoints. The fact that those numbers can be relayed to the inside in a 1:1 fashion is what makes them DID numbers. Of course, a DID number can also point to a hunt pilot, auto-attendant etc.

Now as far as technology goes any trunk that is capable of receiving a called number is capable of DID. SIP trunks and PRIs are both receiving called number while FXO is not. For PRIs you would have to configure "direct-inward-dial" on the incoming pots dial-peer, while there's no need for this with a SIP trunk. For FXOs, since it doesn't get a called number, you have to hard-code the destination with "connection plar opx xxxx" under the voice-port.

So, I'm not exactly sure what you mean by "DID trunk" and "DID service". If you bought a range of numbers from service provider and got a T1 PRI in-place where the carrier is sending those dialed digits, then, by definition, you have DID numbers.

Edit: Unless you are talking about FXS/DID but then the description of it I gave in the post above is still valid.

Edit2: For the sake of correctness, there is a "DID Service" that providers may offer. The provider will assign you a range of numbers but may offer the service of only sending the last 4 or 5 digits so that you don't have to do much digit manipulation. They'd basically do some of your work for you and charge you money (I assume). This may become less and less relevant as folks migrate to the globalized dial-plan.

TheNewITGuy

DID is going to be "Direct Inward Dial" so for example - when a call comes down your service, whether it be PRI/SIP the will be in the form of NPA-NXX-XXX and you decide to translate that number to your internal users with a general 1:1 mapping. In the case of POTS lines, it is a 1:1 map as well, but each "DID" is tied to the individual FXO port unless grouped by the provider.

FXO will connect to the "central office" so a rj-11 from the provider.. T1 port will connect a PRI which is digital where FXO is analog; FXS ports connect to stations that take analog hand offs (fax, wireless phones, etc)

Cisco 2900 can run CME and if that's the case then yes that's your "PBX"

a T1 for data vs T1 for voice are different