Today we are going to continue our conversation about Cisco Voice Gateways. In the beginning we defined what the voice gateways are, and now it is time to see how we can connect our VoIP infrastructure to traditional analog lines or devices. Yes, it is 2022, however the analog connections are still implemented. That’s why you have to know how to use analog voice ports on your voice gateways.
Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire analog circuits in telephony networks. As the name suggests, analog voice ports transport the voice information between devices using analog methods. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs. Also you can use two-wire lines to connect to PBXs or even to PSTN Central Office (CO).
For a voice call to occur, certain information, such as the on-hook status of telephony devices, the availability of the line, and whether an incoming call is trying to reach a device, must be passed between the telephony devices at either end of the call. This information is referred to as signaling. The signaling typically provides information about things such as the following:
- On-hook status
- Ringing
- Line seizure
From signaling prospective there are three types of analog voice interfaces that Cisco voice gateways support. They are shown on the picture below.
A Foreign Exchange Station (FXS) interface connects directly to a standard analog telephone, fax machine, or similar device. It provides ring, voltage, and dial tone to such phones. The Cisco FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and private branch exchanges (PBXs). So in all cases, when you want to connect a traditional analog phone to your Cisco VoIP system, you should use a gateway with an FXS voice port.
That’s how a voice card with FXS ports may look like:
A Foreign Exchange Office (FXO) interface is an RJ-11 connector that allows an analog connection to be directed at the public switched telephone network’s (PSTN’s) central office or to a station interface on a private branch exchange (PBX). The FXO sits on the switch end of the connection. It plugs directly into the line side of the switch so the switch thinks the FXO interface is a telephone.
Let me explain this. Imagine that you have a usual analog PSTN phone and connection in your office. Let’s say the PSTN number of this line and phone is 5551234 (please see the picture below). If you want to allow several employees to use this line for the calls to PSTN, you have to substitute your existing analog phone with a PBX or voice gateway (in case of IP telephony of course). And you connect an FXO port to this PSTN line. In order to work with this line your FXO port should simulate a behavior of a traditional phone.
To make an outbound call to the PSTN through this FXO port, the users of your VoIP network have to dial an access code first (let’s say they dial 9 as the access code). When 9 has been dialed, the FXO port closes the circuit and the user can hear the dial tone and then dial any PSTN number. Of course, only one user can use this line at a time.
To call to your company (inbound calls) the PSTN users will still dial the number of your PSTN line (5551234). The PSTN will send the ringing voltage (AC 90V) to this line, indicating the inbound call to your FXO port. However, the PSTN doesn’t send any dialed number information to the line. So when the FXO port see the inbound call, there is a question, where to send this call (to which internal phone). Typically a system administrator should configure manually a destination for this inbound call at the FXO port (plar configuration in Cisco IOS commands). In other words we say to the system where to route this call – for example, if the call arrive to our FXO port, let’s send it to reception with phone extension 2001.
Below you can find an example of the FXO Cisco voice card.
Cisco FXS and FXO voice ports look very similar to each other!!! Both of them use RJ-11 connector. Please do not mix them up!
E&M Interfaces are trunk circuits to connect telephone switches to one another; they do not connect end-user equipment to the network. The most common form of analog trunk circuit is the E&M (Earth and Magneto) interface, which uses special signaling paths that are separate from the trunk audio path to convey information about the calls. The signaling paths are known as the E-lead and the M-lead. E&M connections from routers to telephone switches or to PBXs are preferable to FXS and FXO connections because E&M provides better answer and disconnect supervision. However, nowadays the E&M interfaces are used very rare and only in case of integration with very old analog PBXs.
E&M is another signaling technique used mainly between PBXs or other network-to-network telephony switches (such as the Lucent 5 Electronic Switching System [5ESS] and Nortel DMS-100). E&M signaling supports tie-line type facilities or signals between voice switches. Instead of superimposing both voice and signaling on the same wire, E&M uses separate paths, or leads, for each.
There are six distinct physical configurations for the signaling part of the interface; they are Types I to V and Signaling System Direct Current No. 5 (SSDC5). They use different methods to signal on-hook or off-hook status, as shown in the following table. Cisco voice implementation supports E&M Types I, II, III, and V.
The physical E&M interface is an RJ-48 connector that connects to PBX trunk lines, which are classified as either two-wire or four-wire (it depends on E&M Type). Two or four wires are used for signaling and the remaining two pairs of wires serve as the audio path. This refers to whether the audio path is full duplex on one pair of wires (two-wire) or on two pair of wires (four-wire).
Here is a Cisco E&M voice port card:
To be continued…
