Step 5: Configuring the PSTN Interface

Step 5: Configuring the PSTN Interface

The next step in Cisco CME system configuration is to route PSTN calls into your office to the extensions on the IP phones, and to allow IP phones to make outgoing PSTN calls. The following sections cover voice port and PSTN trunk configurations necessary to route PSTN calls to IP phones.

Configuring Voice Ports

Chapter 6, "Cisco CME PSTN Connectivity Options," explained the various hardware options you have on Cisco CME for connecting to the PSTN and also provided several configuration examples for different trunk types. The following sections build examples of basic PSTN connectivity for both an analog Foreign Exchange Office (FXO) and a digital T1 Primary Rate Interface (PRI) trunk. It is likely that your office will use one or the other, but not both.

If analog hardware is present in your Cisco CME system, ports show up automatically in the router configuration as follows :

voice-port 2/0/0
voice-port 2/0/1

To add caller ID to analog FXS or FXO ports, use the caller-id command, as shown in Example 15-13.

Example 15-13. Configuring Caller ID on FXS or FXO Ports

cme-3725#

show running-config

voice-port 2/0/0
 caller-id enable
!
voice-port 2/0/1
 caller-id enable

Digital ports do not show up in the configuration simply because the hardware is present in your router. All you see by default is the controller statement alerting you that T1 or E1 port hardware is present. On the newer controller cards that allow software configuration for either T1 or E1 operation on the same hardware, the controller doesn't show up in the configuration until you configure the port type to be either T1 or E1 operation. Use the card type command to accomplish this.

Example 15-14 shows the configuration for a T1 PRI trunk type with the 5ESS switch type. Which switch type you should use depends on the central office you connect to and varies between geographic locations, as well as for T1 compared to E1 ports. The voice port, 0/0:23 in Example 15-13, is created automatically by the pri- group configuration, as is the D channel interface (interface Serial0/0:23 in Example 15-13). If you have T1 Channel Associated Signaling (CAS) or E1 R2 connectivity to the PSTN, you use the ds0-group command instead of the pri-group command.

Example 15-14. Configuring a T1 PRI PSTN Trunk

cme-3725#

show running-config

network-clock-participate wic 0
network-clock-participate aim 0

isdn switch-type primary-5ess

!
controller T1 0/0

pri-group timeslots 1-24

!

interface Serial0/0:23

no ip address
 isdn switch-type primary-5ess
 isdn incoming-voice voice
 no cdp enable
!
voice-port 0/0:23

As covered in Chapter 6, you can use various hardware cards on the router to provide a digital T1 or E1 connection to the PSTN. The configuration shown in Example 15-14, with the exception of the network-clock-participate commands, is generic to all T1/E1 trunks and does not vary based on which hardware you are using.

Routing PSTN Calls to IP Phones

FXO analog lines deliver no dial-in digits (discussed in Chapter 6), so it is necessary to configure an autoterminate destination for these PSTN calls. Ultimately, you should direct these calls to the AA, but because you have not set up the AA for Site A yet, the configuration in this section terminates the calls on extension 2001 for the time being. This is sufficient to test that PSTN calls into your Cisco CME system work properly. In the "Configuring the AA" section later in this chapter, the configuration changes to terminate the PSTN calls onto the AA.

The connection plar opx option does not provide answer supervision (connect) to the PSTN if the Cisco CME IP phone does not answer the call. Thus, it does not generate billing until the call is answered . The connection plar option, on the other hand, generates answer supervision to the PSTN (and therefore starts billing) the moment the router accepts the call, whether or not the call is answered by an IP phone.

The autoterminate destination for calls on the FXO port is configured under the voice-port , as shown in Example 15-15. All calls arriving on voice port 2/0/0 automatically start ringing on extension 2001.

Example 15-15. Configuring FXO Trunks

cme-3725#

show running-config

voice-port 2/0/0

connection plar opx 2001

PSTN calls arriving on the FXO port can now terminate on extension 2001, but calls arriving on the PRI trunk cannot yet ring any phone. All trunks other than FXO provide dialed digits, so the router can switch the calls based on the digits received from the PSTN. However, the digits delivered by the PSTN do not yet match any Cisco CME extension, so the calls receive overflow tone. Some digit manipulation is required. The current ephone-dn for extension 2001 is

ephone-dn 1 dual-line


number 2001

The PSTN number for dialing this phone is 222.555.2001, so you need to change the longer PSTN (E.164) number to 2001 so that it can match the ephone-dn configuration. Digit manipulation can be done in various ways, as explained in Chapter 6. The most straightforward way is simply to configure a secondary number associated with the ephone-dn so that calls to 2001 and calls to 222.555.2001 terminate on the same phone. This is shown in Example 15-16.

Example 15-16. Configuring a Secondary Number

cme-3725#

show running-config

ephone-dn  1  dual-line
 number 2001

secondary 2225552001

Calls coming in from the PSTN to extension 2001 now ring the IP phone and can be answered. You can also use dial plan patterns to accomplish this.

Routing IP Phone Calls to the PSTN

Your employees most likely are used to dialing an access code to get a PSTN line. Assuming that this access code is 9, the dial plan entered into the configuration should direct calls dialed with a leading 9 to the PSTN trunks. Because the Site A system currently has both a PRI and analog FXO trunks to the PSTN, you likely want to give the PRI preference, and use the FXO as a backup only if no timeslots are available. You can achieve this by putting preferences on the dial peers.

The dial plan must also take care of local PSTN calls (9 + seven digits) and long-distance calls (9 + 11 digits). You may also want to add more dial peers to allow (or disallow) international PSTN dialing. The dial plan (and dial peers supporting it) can become very sophisticated. Example 15-17 shows just the basic plain old telephone service (POTS) dial peers necessary on the voice ports to route seven-digit and 11-digit PSTN calls to the PRI trunk first and to the FXO trunk second. The forward-digits command instructs the router to deliver a certain number of digits to the PSTN. In this example, it suppresses the "9" access code and forwards the rest of the digits the IP phone user dialed. You can add any number of digits to or delete any number of digits from the original string, or send completely different digits to the PSTN from what was dialed by the IP phone.

Example 15-17. Configuring POTS Dial Peers for the PSTN

cme-3725#

show running-config

dial-peer voice 1000 pots
 destination-pattern 91..........
 port 0/0:23
 forward-digits 11
!
dial-peer voice 1001 pots
 preference 1
 destination-pattern 91..........
 port 2/0/0
!

dial-peer voice 1002 pots
 destination-pattern 9[2-9]......
 port 0/0:23
 forward-digits 7
!
dial-peer voice 1003 pots
 preference 1
 destination-pattern 9[2-9]......
 port 2/0/0

As soon as you enter this configuration, IP phones can call 914445551212 (11-digit) or 95551212 (seven-digit) and have these calls routed to the PSTN.