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| Section | Weight | Objectives |
|---|---|---|
| Topic 1: Cisco IOS XE Gateway and Media Resources | 15% | - Media resource management - ISDN PRI/BRI configuration and troubleshooting - Voice gateway configuration - MGCP configuration and verification |
| Topic 2: Protocols, Codecs, and Endpoints | 20% | - Codec selection and negotiation - Endpoint deployment and configuration - SIP protocol analysis and troubleshooting - Endpoint troubleshooting |
| Topic 3: Quality of Service (QoS) | 10% | - QoS configuration and monitoring - QoS principles for collaboration - Queuing and policing - Classification and marking |
| Topic 4: Call Control | 25% | - Call routing and toll fraud prevention - Unified Communications Manager architecture - Digit analysis and dial plan design - Call control features and configuration |
| Topic 5: Infrastructure and Design | 20% | - Collaboration architecture design elements - Network components configuration and troubleshooting - Edge devices: Expressway, Cisco Unified Border Element |
| Topic 6: Collaboration Applications | 10% | - Cisco Webex integration - Messaging and presence - Video conferencing deployment |
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NEW QUESTION # 56
Refer to the exhibit.
Cisco Unified Border Element is attempting to establish a call with Subscriber 1, but the call fails. Cisco Unified Border Element then retries the same call with Subscriber 2, and the call proceeds normally. Which action resolves the issue?
Answer: D
NEW QUESTION # 57
Which two models are represented in the Cisco UCM Cluster database? (Choose two.)
Answer: C,D
NEW QUESTION # 58 
Refer to the exhibit. Which route pattern can the Cisco UCM administrator apply to the configuration to allow calls to "1306.1316.1326,13*6.13#6" only?
Answer: D
NEW QUESTION # 59
Refer to the exhibit.
https://i.postimg.cc/C57TkczG/image.png
A call is falling to establish between two SIP Devices The called device answers with these SOP Which SOP parameter causes issue?
Answer: D
Explanation:
The RTP port is used to send and receive media packets during a call. If the RTP port is set to 0, the called device will not be able to send or receive media packets, and the call will fail.
The other options are not correct because:
A). The calling device did not offer a ptime value: The ptime value is used to specify the amount of time between each media packet. If the calling device does not offer a ptime value, the called device will use the default value of 20 milliseconds.
B). The media stream is set to sendonly: The media stream is set to sendonly when the called device is only able to send media packets, and not receive them. This is not a problem, and the call will still succeed.
C). The payload for G.711ulaw must be 18: The payload for G.711ulaw is the type of media packet that is used. The payload must be set to 18 for G.711ulaw, but this is not a problem, and the call will still succeed.
NEW QUESTION # 60
Refer to the exhibit.
https://i.postimg.cc/wMYy0Fhm/image.png
Drag and drop the flow step labels from the left into the correct order on the right to establish this call flow:
* User 1 calls user 2.
* User 2 answers the call
* user 2 disconnects the call
Answer:
Explanation:
Explanation:
1.100 Trying
2.180 Ringing
3.two-way voice path
4.Connect ACK
5.two-way RTP channel
NEW QUESTION # 61
......
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