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Cisco 300-815 Exam (page: 5)
Cisco Implementing Advanced Call Control and Mobility Services (CLASSM)
Updated on: 31-Mar-2026

Viewing Page 5 of 32
300-815 PDF 245 Questions

Users report that calls to some locations do not work. The phone is calling, the recipient picks up, but it does not connect. These users are in the headquarters and registered to the company Cisco UCM with default values for calling (G.711). The problem locations are smaller remote locations with low bandwidth capacity, so the phones are configured with ILBC. Which command should be used to troubleshoot the issue?

  1. show dspfarm dsp active
  2. show dsp active
  3. debug voice dspfarm
  4. debug dsp all

Answer(s): C

Explanation:

When there is a mismatch in codec capabilities -- such as G.711 at headquarters and iLBC at remote sites -- a transcoder is needed. To troubleshoot codec negotiation and transcoding issues, the command debug voice dspfarm provides detailed real-time information about DSP resource usage and media processing, which is crucial in identifying why calls fail to connect despite being answered.



An engineer receives reports from users that calls to external participants on mobile phones are failing. The gateway is based on a server client setup.
When logs are analyzed, what is the last message expected before the media is established that Cisco UCM sends?

  1. established connection with an early SDP to the gateway, which must respond with 200 OK
  2. modified connection with SDP to the gateway, which must respond with 200 OK
  3. established connection with SDP to the gateway, which must respond with 200 OK
  4. forward connection with an early offer to the gateway, which must respond with 200 OK

Answer(s): C

Explanation:

In a server-client gateway setup, Cisco UCM sends an INVITE with SDP (an offer) to the gateway to establish the media path. The last signaling message from Cisco UCM before media is established is the INVITE with SDP, and the expected response from the gateway is 200 OK, confirming that the call setup is complete and media can flow.



Users of a Cisco UCME report that calls lose audio after they are forwarded to some users. Calling point-to- point seems to function fine, but depending on the type of phone, some calls lose audio when forwarded.
Looking at the trace header that shows Allow:

INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER

What is missing and is a reason that the SRTP stream stops working?

  1. EARLY-OFFER
  2. FORWARD
  3. UPDATE
  4. REDIRECT

Answer(s): C

Explanation:

The UPDATE method is used in SIP to modify session parameters (such as media) before a final response (like 200 OK) is sent. If the Allow header on a SIP INVITE does not include UPDATE, then SIP endpoints or gateways cannot renegotiate media parameters mid-dialog -- such as when a call is forwarded.

Without support for UPDATE, SRTP streams cannot be correctly re-established or modified during call forwarding, resulting in loss of audio.



An engineer must verify that a Cisco 8865 telephone is advertising unsolicited NOTIFY with out-of-band DTMF relay for SIP. The solution is using Cisco UCME and Cisco Unified Border Element. How must the engineer look for this confirmation?

  1. On IOS, run the debug ccsip all command and look for "Call-Info…..method="NOTIFY ;Event=telephone-event" inside the NOTIFY message.
  2. On IOS, run the debug ccsip dtmf command and look for "Call-Info…..method="NOTIFY ;Event=telephone-event" inside the NOTIFY message.
  3. On IOS, run the debug ccsip messages command and look for "Call-Info…..method="NOTIFY ;Event=telephone-event" inside the INVITE message.
  4. On IOS, run the debug ccsip all command and look for "Call-Info…..method-'NOTIFY ;Event=DTMF" inside the INVITE message.

Answer(s): A

Explanation:

To verify that the Cisco 8865 phone is using unsolicited NOTIFY with out-of-band DTMF relay, the engineer must capture and inspect SIP messages. The debug ccsip all command provides detailed SIP signaling, and the presence of "Call-Info: ... method="NOTIFY"; event=telephone-event" inside the NOTIFY message confirms that DTMF is being sent out-of-band using NOTIFY, as expected.



A company is using a H.323 gateway for PSTN calls from Cisco phones connected to a Cisco UCM cluster. There are reports of problems getting audio on these incoming calls from the PSTN. The phone rings, but nothing is heard and the call disconnects.
Which command troubleshoots this issue?

  1. debug h245 asn1
  2. debug h245 pstn
  3. debug voice h323 inout
  4. debug voice sip inout

Answer(s): C

Explanation:

The debug voice h323 inout command is used to troubleshoot H.323 call signaling and media negotiation issues between Cisco Unified CM and an H.323 gateway. It provides visibility into call setup, media channel establishment, and disconnect causes, which is essential when calls ring but fail to establish audio and then drop.



Which top-level IOS command is needed to begin the configuration of a Cisco Unified Communications Manager Express gateway to enable phones to be registered via SIP?

  1. allow-connections sip to sip
  2. voice service voip
  3. voice register global
  4. voice register dn

Answer(s): C


Reference:

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications- manager-express/99946-cme-sip-guide.html



An IP Telephony administrator is deploying IP phones. The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111.2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two.)






Answer(s): D,E



An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor incoming command configured? (Choose two.)

  1. "voice register pool " configuration mode
  2. "ephone-dn " configuration mode
  3. "dial-peer cor custom " configuration mode
  4. "voice register global " configuration mode
  5. "telephony-service " configuration mode

Answer(s): A,B



Viewing Page 5 of 32



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