Here you have the best Cisco CLASSM practice exam questions
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.)
During a SIP call transfer, the transferring phone (Phone A) sends a SIP-REFER message to the Call Manager with the information for the transfer target (Phone C). Therefore, Phone A sends a SIP-REFER message. Secondly, when Phone A initiates the transfer by pressing the Transfer button, Phone B will hear Music on Hold (MOH). The MOH audio source is typically determined by the network hold settings of the transferring phone, which in this case is Phone A. Therefore, the correct scenarios in the exhibit are that Phone A sends the SIP-REFER message to the Call Manager, and Phone B hears the MOH chosen from Phone A's Network Hold MOH Audio Source settings.
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?
No DTMF is negotiated. The exhibit shows the SDP (Session Description Protocol) information for the SIP call setup. For DTMF to work in-band, it's essential that the DTMF tones are negotiated and supported. In this case, the RTP payload types 18 for the codec and 110 for DTMF are listed, but there is no evidence showing an actual negotiation that ensures the DTMF tones will transmit correctly over the network. Therefore, the issue arises due to the lack of DTMF negotiation.
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 one-way 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).
To resolve a one-way audio issue in an H.323 protocol call using slow-start mode, it's essential to locate the Real-Time Transport Protocol (RTP) IP and port information. This information can be found in the H.245 Open Logical Channel Acknowledgment (Ack) message. The H.245 Open Logical Channel message initially requests the opening of a communication channel, but it is in the H.245 Open Logical Channel Ack where the endpoint confirms and provides the RTP IP and port details that will be used for the media. Therefore, the correct option is H.245 Open Logical Channel Ack.
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
For fast start-to-early media scenarios in H.323 to SIP interworking, BFCP (Binary Floor Control Protocol) and AUDIO capabilities must be configured on dial peers. BFCP is needed for content control during the media session, and AUDIO allows for early media transmission, enabling users to hear audio before the call is fully established. These are essential for optimizing call setup times and ensuring a smooth user experience.
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
When troubleshooting H.323 call setup, the 'ALERTING' message informs you that the called party is being notified about the call. This message is sent once the call setup process has reached the called party, indicating that the remote endpoint has initiated the process to alert the user, usually by ringing the phone.