You issue the show running-config command on a CME router and receive the following partial output:

Examine the output, and use the information you gather to answer the question.
Which of the following best describes port 1/0/0 in this configuration? (Select the best answer.)

A.
an analog FXO port

B.
an analog FXS port

C.
a SIP trunk port

D.
a VoIP port

Explanation:
Port 1/0/0 is an analog foreign exchange office (FXO) port. An FXO interface is typically used to connect an
analog device to the public switched telephone network (PSTN). In addition, if a public branch exchange (PBX)
is configured with a foreign exchange station (FXS) port, the FXO interface on an analog device can terminatean analog trunk line from a PBX. FXO interfaces are commonly found on standard telephones, fax machines,
and analog modems.
The port command is used by a voice router to match inbound plain old telephone service (POTS) dial peers
and to determine where to route outgoing POTS dial peers. The dial peer voice command is used to define how
calls are routed to destination endpoints on either the PSTN or a Voice over IP (VoIP) network. To define call
routing for the PSTN, you should issue the dialpeer voice command with the pots keyword. To define call
routing for a VoIP network, you should issue the dialpeer voice command with the voip keyword. In this
scenario, dial peer 1 is configured as a pots dial peer. Therefore, the Cisco Unified Communications Manager
Express (CME) router port that is connected to the PSTN is FXO port 1/0/0.
Port 1/0/0 is not an analog FXS port. An analog trunk line from the central office (CO) typically originates from
an FXS interface on a phone switch. The switch provides dial tone, ring voltage, and line voltage for the
customer site. Because the FXS interface on the phone switch provides power, it cannot be connected to
another FXS interface; instead, the FXS interface must be connected to a device with an FXO interface, such
as an analog telephone or a legacy voice mail system.
Port 1/0/0 is neither a Session Initiation Protocol (SIP) trunk port nor a VoIP port. SIP is the signaling method
that is most commonly used by Internet telephony service providers (ITSPs). ITSPs enable customers to use
VoIP to make phone calls over the Internet. SIP is an Internet Engineering Task Force (IETF)standard call
signaling protocol that is supported by a wide variety of IP telephony vendors. The configuration methods for
SIP trunking among ITSPs vary. To configure a CME dial peer for SIP trunking in Cisco IOS, you should issue
the session protocol sipv2 command in dial peer configuration mode. For example, the following configuration
creates dial peer 5012 to handle an outgoing call to a SIP trunk connected to the PSTN:
dialpeer voice 5012 voip
session protocol sipv2
session target ipv4:10.11.12.13
dtmfrelay sipnotify
no vad
The configuration above sets the trunking protocol for the dial peer to SIP version 2, identifies the SIP trunk as
having the IP address 10.11.12.13, specifies that dualtone multifrequency (DTMF) dialed digits should be
relayed through SIP’s NOTIFY messages, and disables voice activity detection (VAD) for the dial peer.
Because no codecconfiguration command is issued, the default G.729 codec will be used for the dial peer.

https://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/
vrht_d1_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1458170
https://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/
vrht_p2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1107545