Where is the dtmf-relay command configured on Cisco Unified Border Element?

Where is the dtmf-relay command configured on Cisco Unified Border Element?A . in the voice-class VoIP configurationB . in the VoIP dial peerC . in global SIP configurationD . in the VoIP or POTS dial peersView AnswerAnswer: B Explanation: Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/dtmf­relay.html

May 4, 2020 No Comments READ MORE +

Which action must the administrator take to fix the issue?

Refer to the exhibit. Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number “222333444” and Cisco Unified Communications Manager is expecting the called number...

May 4, 2020 No Comments READ MORE +

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?

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?A . allow-connections sip to sipB . voice service voipC . voice register globalD . voice register dnView AnswerAnswer: C Explanation: Reference: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified­communications-manager-express/99946-cme-sip-guide.html

May 3, 2020 No Comments READ MORE +

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?

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?A . The negotiated RTP port is outside of the range described by RFC, so...

May 3, 2020 No Comments READ MORE +

To see signaling for media and call setup, which debug must the Administrator turn on?

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?A . debugB . 323 messagesC . debugD . 225 asn1E . debugF . 246 asn 1G . debugH . 225 media ....

May 3, 2020 No Comments READ MORE +

Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)A . DTMFB . BFCPC . VIDEOD . FAXE . AUDIOView AnswerAnswer: AB

May 2, 2020 No Comments READ MORE +

What are two possible solutions?

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...

May 2, 2020 No Comments READ MORE +

Which rule modified DNIS in the format that the provider is expecting?

Refer to the exhibit. Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element...

May 2, 2020 No Comments READ MORE +

Where can you find the RTP IP and port information for both sides?

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...

May 1, 2020 No Comments READ MORE +

Which description of RTP timestamps or sequence numbers is true?

Which description of RTP timestamps or sequence numbers is true?A . The sequence number is used to detect losses.B . Timestamps increase by the time “carrying” by a packet.C . Sequence numbers increase by four for each RTP packet transmitted.D . The timestamp is used to place the incoming audio...

May 1, 2020 No Comments READ MORE +