Top 50 Cisco IPT Interview Questions and Answers (2026 Guide)

Cisco IPT Interview Questions and Answers – So, you are aiming for a job as a Cisco IP Telephony engineer? We have spent the last decade designing and troubleshooting Cisco Unified Communications Manager (CUCM) for enterprises. We know exactly what interviewers look for.

Based on real-world configuration guides and troubleshooting scenarios, We have compiled 50 Cisco IPT Interview Questions and Answers. This guide covers everything from basic clustering to advanced dial plan manipulation. Let’s get you hired.

Table of Contents

Part 1: Clustering & Deployment Models (Q 1-10)

These questions test your knowledge of how CUCM servers talk to each other.

1. What is a CUCM Cluster?

A cluster is a set of CUCM servers that share the same database and resources. It acts as a single call processing agent. You can have only one Publisher (read/write DB) per cluster and multiple Subscribers (read-only DB for redundancy).

2. Explain the difference between Single Site and Multisite Centralized deployment.

Single Site: One cluster in one location. All phones use high-bandwidth codecs (G.711). If the WAN is down, the PSTN handles everything.
Centralized: One cluster in a data center supports remote branch phones over the WAN. Remote sites use SRST (Survivable Remote Site Telephony) to make calls if the WAN fails.

3. What is the maximum one-way delay for Clustering over the WAN?

To avoid call setup failures, the maximum one-way delay between any two CUCM servers must not exceed 40 milliseconds (or 80 ms round-trip).

4. What is the difference between a Region and a Location?

Region: Defines which codec (G.711 vs G.729) is used between two devices.
Location: Defines the bandwidth limit for a WAN link for Call Admission Control (CAC).

5. What is a Media Resource Group List (MRGL)?

It is a prioritized list of resources (like conference bridges or MTPs). It tells CUCM which hardware resource to use first (e.g., use the local DSP before using a software resource).

6. How does an IP Phone get its configuration?

Phone powers on (PoE).
Gets VLAN via CDP.
Requests DHCP (looks for Option 150).
Downloads config file from TFTP.
Registers to CUCM.

7. What is the role of the Publisher in CUCM?

The Publisher is the “first node.” It has the only read/write copy of the SQL database. Subscribers replicate from it. If the Publisher is down, call processing continues, but you cannot make admin changes.

8. What is H.323 vs MGCP?

H.323: Peer-to-peer. The gateway has intelligence (dial-peers). Harder to configure but feature-rich.
MGCP: Master/slave. CUCM controls every port on the gateway. Easier dial plan (no dial peers on the GW), but the gateway is dumb.

9. What is the “PRI Backhaul” in MGCP?

It is when the gateway terminates ISDN Layer 2 (Q.921), but forwards Layer 3 (Q.931) signals back to CUCM via TCP port 2428. CUCM handles the call routing.

10. What is a Gatekeeper?

An H.323 entity that provides address translation (E.164 to IP) and bandwidth management for H.323 endpoints and gateways.

Part 2: Dial Plan Design (Q 11-20)

Dial plans are the heart of Cisco IPT Interview Questions and Answers.

11. What is a Partition and Calling Search Space (CSS)?

Partition: A logical group of route patterns or directory numbers (like a “folder”).
CSS: An ordered list of Partitions. It defines what a phone is allowed to dial.

12. Scenario: Line CSS vs. Device CSS. Which wins?

Answer: The intersection wins (most restrictive).

Line CSS (User level) + Device CSS (Phone level).
CUCM looks at both. If the Line CSS allows “Local” but Device CSS blocks “Long Distance,” the call is blocked.

13. What is a Translation Pattern?

It changes digits before routing. Unlike a route pattern, it does not select a gateway; it just manipulates digits and sends the call back for a second digit analysis.

14. Scenario: Non-Urgent Translation Patterns.

In CUCM 7.x+, if you configure a pattern for 9011. (international) and set it to Urgent, the system matches the first digit and routes immediately. This breaks variable-length dialing. You must set international patterns to Non-Urgent.

15. What is a Local Route Group (LRG)?

A feature that simplifies TEHO (Tail End Hop Off). It automatically uses a gateway local to the called party without building hundreds of explicit route lists.

16. How does Digit Manipulation work on a Gateway?

Using commands like prefix, forward-digits, or translation-profile to strip access codes (like 9) or add area codes.

17. Scenario: Overlapping Numbering Plans.

Two sites both have extension “1001.” How do they call each other?
Answer: You implement Site Code dialing. The user dials an Intersite Prefix (e.g., 8) + Site Code (e.g., 81) + Extension (1001). The gateway strips the 881 before sending to the remote site.

18. What is a Hunt List?

A configuration that allows calls to roll over to different line groups. For example: Ring Group A for 10 seconds, then Group B.

19. What is the difference between Ad Hoc and Meet-Me conferencing?

Ad Hoc: An active user adds participants one by one (starts as a 2-way call, becomes 3-way).
Meet-Me: Users dial a specific conference bridge number at a scheduled time.

20. What is a Route Pattern “@” (At-sign)?

It uses a Route Filter to match the North American Numbering Plan (NANP) patterns automatically (e.g., distinguishing 7-digit local from 11-digit long distance).

Part 3: Voice Gateways & Protocols (Q 21-30)

21. Scenario: My H.323 Gateway shows “Unknown” in CUCM. Why?

Answer: H.323 is peer-to-peer. It does not “register” like MGCP. Ensure the gateway IP is correct, and verify the dial-peer session target points to CUCM. Also, check that the Gatekeeper (if used) is resolving the alias.

22. What is the difference between FXS and FXO?

FXS (Wall Plug): Provides dial tone, battery, and ring voltage to a phone.
FXO (Phone Plug): Receives dial tone (connects to the wall). You connect an FXO port to an FXS port.

23. What ports does MGCP use?

UDP 2427: Call Control (Gateway <-> CUCM).
TCP 2428: Keepalives and PRI Backhaul.

24. Scenario: “show isdn status” shows Layer 2 as “TEI_ASSIGNED” not “MULTIPLE_FRAME_ESTABLISHED.”

Answer: Layer 2 is down. This usually means:

Wrong isdn switch-type configured.
Losing clocking on the line.
The Telco side is down. Check with the provider.

25. What is a Transcoder (XCODE)?

A DSP resource that converts a stream from one codec to another (e.g., G.711 on the LAN to G.729 on the WAN).

26. How do you ensure a conference uses G.711 and not G.729?

Place the conference bridge resource in a Region that is configured for G.711 to the phones. The Region setting overrides the codec negotiation.

27. What is a SIP Trunk?

A connection between CUCM and a SIP provider or another PBK using Session Initiation Protocol. It uses port 5060 (UDP or TCP).

28. Scenario: DTMF tones (IVR menus) fail across a SIP trunk.

Answer: Low-bandwidth codecs (G.729) destroy DTMF fidelity. Enable DTMF Relay. On SIP trunks, use dtmf-relay rtp-nte or sip-notify to send digits out-of-band.

29. What is the default incoming dial-peer in IOS?

POTS dial-peer 0 (zero). It is used when no specific incoming called-number matches. Usually, we configure a POTS dial-peer with incoming called-number . to catch all calls.

30. What is the purpose of “Call Preservation” in MGCP?

If the CUCM connection drops, MGCP maintains existing calls so users don’t get disconnected during a failover.

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Part 4: Media Resources & QoS (Q 31-40)

31. What is MTP (Media Termination Point)?

A resource that bridges H.245 streams. It is required for H.323 v1 endpoints to hold calls or for SIP to SCCP DTMF interworking.

32. Scenario: Software Conference Bridge limitations.

Software bridges (Cisco IP Voice Media Streaming App) only mix G.711 streams. If you have G.729 participants, you must insert a Transcoder before the conference or use hardware DSPs.

33. What is Music on Hold (MOH) and the “G.729 quality issue”?

MoH streams music. G.729 is optimized for human speech, not music. It makes MoH sound “tinny.” Solution: Put the MOH server in a separate Region so MoH streams G.711 while voice stays G.729.

34. What is DSCP for CUCM signaling?

Intra-cluster traffic (ICCS) uses IP Precedence 3 (DSCP 24). Voice RTP payload uses DSCP 46 (EF).

35. Scenario: The phone works, but the “Hold” button doesn’t play music.

Answer: Check the MOH Audio Source. If the server is in “Down” state or no files are uploaded, it will play silence. Also, verify the MRGL assigned to the device includes the MOH server.

36. What is the difference between Barge and Privacy?

Barge: Allows a user to join a shared line call in progress.
Privacy: Blocks other phones with the shared line from seeing or barging your call.

37. How do you configure a Hardware Conference Bridge?

Configure the router with dspfarm.
Register it to CUCM via SCCP using sccp local and sccp ccm.
Add the device to CUCM as a “Conference Bridge.”
Add it to an MRGL.

38. What is Extension Mobility (EM)?

Users log into any phone in the cluster and their personal profile (line appearances, speed dials) loads onto that phone.

39. Scenario: Extension Mobility login fails.

Answer: Check the User Device Profile. The profile must be assigned to a compatible phone model (e.g., a 7960 profile won’t load on a 7940). Also, verify the user has a PIN configured, not just a password.

40. What is Mobile Voice Access (MVA)?

Allows a cell phone user to dial into CUCM, enter a PIN, and get an internal dial tone. They can then make business calls that appear to come from their desk phone.

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Part 5: Advanced Scenarios (Q 41-50)

41. Scenario: Call Park vs. Directed Call Park.

Scenario: A receptionist parks a call, but the CEO retrieves the wrong call.
Answer: Use Directed Call Park. The admin configures a retrieval prefix (e.g., *51*). The user must dial the park slot number (e.g., 01) to get the call. This prevents accidental pickups.

42. Scenario: SRST Failover not working.

The WAN is down, but SCCP phones show “Registering” and never go to SRST.
Answer: Check the SRST Reference in the Device Pool. Ensure the ip source-address on the router matches the SRST Reference IP. Also, ensure call-manager-fallback is configured.

43. Scenario: Inbound POTS call drops immediately.

Answer: It likely failed Inbound Dial Peer matching. Use debug voip ccapi inout. If no dial peer is matched (service using default dial-peer 0), the call treats it as a modem call and drops. Create a dial-peer with incoming called-number . and direct-inward-dial.

44. How do you check PRI Layer 3 status?

show isdn status -> Look for “Layer 3 Status: Active Layer 3 Call(s)“. If it is 0, but the phone number is good, Layer 3 is down.

45. Scenario: H.323 Gateway no voice path (One-way audio).

Answer: H.323 often has routing issues. Ensure no ip nat inside on the voice VLAN interface. If NAT is involved, you need an MTP to rewrite the IP addresses in the H.245 headers.

46. What is COR (Class of Restriction)?

A feature on IOS gateways (like SRST) that acts as a “Lock and Key.” The incoming dial peer has a “Key” (COR list), and the outgoing dial peer has a “Lock.” The call only proceeds if the Key unlocks the Lock.

47. Scenario: Implementing “Lobby Phone” restrictions.

Answer: Create a new Partition for “Local_Calls.” Put the Lobby Phone’s CSS in a Partition that only contains local route patterns. Remove the Partition that contains “Long Distance” (91[2-9]…).

48. What is the “Music on Hold” active vs network hold?

Active Hold: User presses the Hold button (User hears MoH).
Network Hold: The call is parked or transferred (Caller hears MoH).

49. Scenario: Video call fails over inter-cluster trunk.

Answer: Video often requires high bandwidth. Verify the Locations Bandwidth (set to 384k+). Also, note that the Cisco VT Camera Wideband codec is not supported over inter-cluster trunks (ICT).

50. Final Scenario: The “One-Way Audio” checklist.

You are on site and a user cannot hear the PSTN caller, but the PSTN hears them.
Step 1: Check the Region configuration. Is the codec match correct?
Step 2: Check MTP Required on the Trunk. Is it checked unnecessarily?
Step 3: Check IP Routing. Does the voice gateway have a route back to the phone subnet?
Step 4: Check NAT. Is the gateway sending private IPs in the SDP?

Conclusion

Mastering these Cisco IPT Interview Questions and Answers requires knowing the theory and the “why” behind the commands. Focus on the scenarios—most failures are due to Dial Peer mis-matches, Region codec mismatches, or MGCP registration details.

If you understand the difference between a Publisher and Subscriber, and when to use a Transcoder versus an MTP, you are already ahead of 80% of candidates. Good luck with your Cisco IPT interview

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