A voice gateway has a POTS dial peer connected to a legacy PBX via a T1/E1 interface. The PBX sends called numbers in a 4-digit format, but the gateway must route calls to the PSTN using a 10-digit format. What should you configure on the POTS dial peer?

Use the 'num-exp' command to expand the number globally.

Configure a destination pattern that matches the 4-digit number and use a translation rule to prepend the area code.

Set the port to the T1/E1 interface and configure the signaling to match the PBX.

Apply a voice translation profile to the POTS dial peer to expand the called number to 10 digits.

What is the purpose of options keepalive on a Cisco voice gateway?

To set the destination pattern for a dial peer

To specify the DTMF relay method for a call

To monitor server availability and trigger failover

To control the codec used for a call

A voice gateway connects to a SIP trunk that supports G.711 and G.729, but the gateway's local network uses G.722 for internal calls. You want to prefer G.711 for calls to the SIP trunk, but allow G.729 if G.711 is not available. What should you configure on the VoIP dial peer?

Use the 'voice-class codec' command to set the codec to G.722.

Configure the dial peer with 'codec g711ulaw' and 'codec g729r8' as separate commands.

Set the codec to G.711 only on the dial peer.

Define a codec preference list that includes G.711 first, then G.729, and apply it to the VoIP dial peer.

A voice gateway is configured with a VoIP dial peer to route calls to a SIP provider. The provider only supports G.729 and requires DTMF relay via RFC 2833. You need to ensure the dial peer uses G.729 and relays DTMF correctly. What should you configure on the VoIP dial peer?

Configure the dial peer with 'codec g729r8' and 'dtmf-relay cisco-rtp'.

Enable 'dtmf-relay sip-kpml' and set the codec to G.711.

Set the codec to G.729 and enable 'dtmf-relay rtp-nte'.

Create a codec preference list with G.729 and apply it, but leave DTMF as default.

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Domain 4Objective 1

4.1 Configure Voice Gateway and Session Border Controller Elements for On-Premises and Cloud Calling 350-801 Practice Questions (Page 5)

Part of the Voice Gateways and Session Border Controllers domain, which accounts for 10% of the 350-801 exam. Cisco does not publish an official question count, but from its 120-minute exam (~50–80 total, ~5–8 in this domain), expect 2–3 from this objective — we provide 27 practice questions to prepare you well beyond it. (estimate)

27questions here

6free pages

9concepts

10%of the exam

Questions 21–25

21application · medium
A voice gateway has a POTS dial peer connected to a legacy PBX via a T1/E1 interface. The PBX sends called numbers in a 4-digit format, but the gateway must route calls to the PSTN using a 10-digit format. What should you configure on the POTS dial peer?
Select an answer first
22foundation · easy
What is the purpose of options keepalive on a Cisco voice gateway?
Select an answer first
23application · medium
A voice gateway connects to a SIP trunk that supports G.711 and G.729, but the gateway's local network uses G.722 for internal calls. You want to prefer G.711 for calls to the SIP trunk, but allow G.729 if G.711 is not available. What should you configure on the VoIP dial peer?
Select an answer first
24foundation · easy
Which command is used to create a voice translation rule on a Cisco IOS voice gateway?
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25application · medium
A voice gateway is configured with a VoIP dial peer to route calls to a SIP provider. The provider only supports G.729 and requires DTMF relay via RFC 2833. You need to ensure the dial peer uses G.729 and relays DTMF correctly. What should you configure on the VoIP dial peer?
Select an answer first

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