Step 5: Configuring the PSTN Interface
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
voice-port 2/0/0 voice-port 2/0/1
To add caller ID to analog FXS or FXO ports, use the
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
statement alerting you that T1 or E1 port hardware is present. On the
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
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.
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
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
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
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
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
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.