10 free Cisco 300-815 CLACC practice questions with the correct answer and a full explanation for each, taken from the CertStash pack of 245 questions. Work through them, then open each answer to check your reasoning.
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Question 1
Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)
Show answer and explanation
Correct answer: A, D
A. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section. D. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings. When Phone A initiates a blind transfer by pressing the Transfer button, Phone A sends the SIP REFER message to the CUCM (not Phone B), making option A correct. When Phone A puts Phone B on hold by pressing Transfer, Phone B immediately hears music on hold. The MOH source is determined by Phone A's Network Hold MOH Audio Source settings, not Phone B's User Hold settings and not Phone A's User Hold settings. The Network Hold MOH Audio Source is used when a call is placed on hold during transfer or other network-initiated hold scenarios, making option D correct.
Why the other options are wrong
- B. Phone B does not send the SIP REFER message; Phone A (the initiating party) sends it to the CUCM.
- C. The MOH audio source is determined by the holding party's (Phone A's) Network Hold settings, not Phone B's User Hold settings.
- E. The MOH audio is chosen from Phone A's Network Hold MOH Audio Source settings during transfer, not from User Hold settings.
Question 2
Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?
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Correct answer: D. No DTMF is negotiated.
The SIP 200 OK response shows media negotiation for the call. In the first SDP (offer), the payload type 110 is defined as 'a=rtpmap:110 telephone-event/8000', which is the standard in-band DTMF mechanism. However, in the ACK response from the other endpoint, the media line shows 'm=audio 8056 RTP/AVP 18' with only payload type 18 (G729) negotiated. Critically, there is no corresponding 'a=rtpmap:110 telephone-event/8000' in the ACK response, meaning the telephone-event payload type was not accepted by the far end. Since only codec 18 is negotiated and no DTMF mechanism is agreed upon in the final SDP, DTMF digits cannot be transmitted in-band, causing the inability to enter digits.
Why the other options are wrong
- A. RFC standards define RTP ports flexibly; the port numbers 8190 and 8056 are within valid ranges and not the cause of DTMF failure.
- B. The SDP exchange shows Early Offer negotiation occurring (offer in 200 OK, answer in ACK), which is consistent with DTMF support; this is not a Delayed Offer scenario.
- C. The rtpmap:0 refers to PCMU codec, not DTMF; the DTMF issue is that rtpmap:110 for telephone-event is missing from the ACK, not that payload 0 is marked inactive.
Question 3
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the on-ay audio call.
You gather the H.225 and H.245 messages for one of the one-way audio calls.
Where can you find the RTP IP and port information for both sides?
(Note: This call flow has not invoked any media resources like MTP or transcoders).
Show answer and explanation
Correct answer: D. H.245 Open Logical Channel Ack
In H.323 slow-start mode, the RTP IP and port information for both sides of a call is communicated in the H.245 Open Logical Channel Ack message. The originating endpoint sends an Open Logical Channel message proposing media parameters, and the responding endpoint confirms the agreed-upon RTP parameters (including IP address and port) in the Open Logical Channel Ack. This acknowledgment contains the actual media endpoint details that both sides will use for RTP transmission.
Why the other options are wrong
- A. Terminal Capability Set messages describe what codecs and capabilities an endpoint supports, not the actual RTP transport addresses.
- B. The Open Logical Channel message is a proposal; it does not contain confirmed RTP endpoint information from both sides.
- C. H.225 Connect is a call signaling message that does not carry the detailed RTP transport address information negotiated during H.245 media setup.
Question 4
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
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Correct answer: A, D
A. DTMF D. FAX For fast start-to-early media scenarios in H.323 to SIP interworking, AUDIO and FAX extended capabilities must be configured on dial peers. These capabilities ensure that media streams (both voice and fax) are properly negotiated and established during the early media phase before the call is fully connected, allowing media to flow before answer. DTMF, BFCP, and VIDEO are either signaling methods or optional codecs that are not required for basic fast start early media scenarios.
Why the other options are wrong
- B. BFCP (Binary Floor Control Protocol) is for conference floor control and is not a required extended capability for fast start early media.
- C. VIDEO is an optional codec capability and not a required extended capability for fast start early media in standard H.323 to SIP scenarios.
- E. AUDIO alone is insufficient; FAX capability must also be configured to handle fax media in interworking scenarios.
Question 5
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
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Correct answer: A. ALERTING
The ALERTING message in H.323 call setup informs that the called party is being notified about the incoming call. This message is sent by the called endpoint to indicate that the user is being alerted (typically causing the phone to ring) but the call has not yet been answered. ALERTING represents the state where the remote party's device is notifying the user of the call.
Why the other options are wrong
- B. PROCEEDING indicates that the call setup is progressing but does not specifically indicate that the called party is being notified.
- C. CONNECT is sent after the called party has answered the call, not when they are being notified of it.
- D. RINGING is not a standard H.323 message; ALERTING is the correct H.323 message used to indicate the ringing state.
Question 6
End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
Show answer and explanation
Correct answer: D. c= line of SDP content
The c= (connection) line in SDP content specifies the media connection address where RTP traffic should be sent. When troubleshooting one-way audio issues in SIP calls that work with PSTN but not between internal sites, the c= line provides critical information about where the endpoint expects to receive media. Mismatches or incorrect IP addresses in the c= line are a common cause of one-way audio, making it the key field to examine for this symptom.
Why the other options are wrong
- A. The Contact header in a 200 OK response identifies where to send future SIP requests, not the media connection address for RTP.
- B. The Allow header lists which SIP methods the endpoint supports and does not provide media connection information.
- C. The o= (origin) line identifies the session originator and version information but does not specify the media connection address where RTP should be sent.
Question 7
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
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Correct answer: A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
If the far end connection data (c=) in the SDP is overwritten by deep packet inspection or modification in the call signaling path, the originating endpoint will send RTP traffic to the wrong destination IP address. Deep packet inspection devices or middle boxes that modify SDP content can cause the connection address to be changed, resulting in RTP packets being sent to an address where the receiving endpoint is not actually listening, causing complete media failure.
Why the other options are wrong
- B. When Cisco Unified Communications Manager invokes media termination point resources, those resources become part of the media path intentionally and handle the RTP traffic appropriately.
- C. RTP traffic arriving beyond the jitter buffer would result in dropped packets and degraded audio quality, not a complete failure to receive RTP from the originating endpoint.
- D. RTP uses UDP ports in the range 16384-32768, not TCP; blocking TCP ports 1638-2768 would not prevent UDP RTP traffic from being received.
Question 8
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?
Show answer and explanation
Correct answer: B. debug H.225 asn1
The debug H.225 asn1 command displays H.225 messages in ASN.1 format, which includes both call setup signaling and media-related information. H.225 handles the call control and signaling aspects of H.323, including the initial setup and alerting messages. To see both media and call setup signaling on an H.323 gateway via CLI, debugging H.225 ASN.1 provides the necessary visibility into the complete call flow.
Why the other options are wrong
- A. There is no standard command "debug H.323 messages"; H.323 debugging must be more specific to H.225 or H.245 protocols.
- C. H.246 does not exist in H.323 protocol specifications; the correct protocols are H.225 and H.245.
- D. There is no standard command "debug H.225 media"; media debugging would be under H.245, not H.225.
- E. While debug H.323 asn1 may be a valid command on some platforms, debug H.225 asn1 is the more direct and standard method for viewing call setup signaling.
Question 9
What is first preference condition matched in a SIP-enabled incoming dial peer?
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Correct answer: A. incoming uri
The incoming URI is the first preference condition matched in a SIP-enabled incoming dial peer. When a SIP INVITE arrives, the system first attempts to match the Request-URI against the incoming URI pattern configured on dial peers. This is the highest priority matching criterion for routing incoming SIP calls to the appropriate dial peer.
Why the other options are wrong
- B. Target carrier-id is a lower priority matching criterion used when URI-based matching is not available or does not match.
- C. Answer-address is not a standard incoming dial peer matching criterion for SIP calls.
- D. Incoming called-number is a lower priority criterion; the incoming URI takes precedence in SIP dial peer matching.
Question 10
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally.
What are two possible solutions? (Choose two.)
Show answer and explanation
Correct answer: C, D
C. Ask the firewall administrator to change the range of UDP ports to 1638-2767. D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000. The intermittent one-way audio issue is caused by a firewall restriction on UDP media ports. Since the firewall only allows UDP ports 20000-22000 but RTP traffic uses the standard range 16384-32767, packets outside the allowed range are being blocked. Two solutions are: (1) ask the firewall administrator to open the correct UDP port range 1638-2767 to match RTP requirements, or (2) configure the SIP profile in Cisco Unified CM to restrict media ports to the firewall-allowed range 20000-22000. Changing to TCP is incorrect because RTP uses UDP, and changing system parameters instead of the SIP profile is incorrect because the profile is the appropriate configuration level for media port ranges.
Why the other options are wrong
- A. Changing the SIP profile media port range to 16384-32767 would exceed the firewall's allowed range, not solving the problem.
- B. RTP uses UDP, not TCP; changing the firewall protocol would not resolve the issue.
- E. System Parameters is not the correct location to configure media port ranges; the SIP profile assigned to the phones is the appropriate configuration level.
That was 10 of 245.
The full Cisco 300-815 CLACC pack has all 245 questions, each with the answer, the explanation and why the other options are wrong, plus a questions-only copy for timed runs. US$39, paid once, with free monthly updates and a pass-or-your-money-back guarantee.