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Exam 300-815 topic 1 question 31 discussion

Actual exam question from Cisco's 300-815
Question #: 31
Topic #: 1
[All 300-815 Questions]

After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

  • A. Router(config)# voice service voip Router(conf-voi-serv)#allow-connections h323 to h323
  • B. Router(config)#dial-peer voice 2 voip Router(config-dial-peer)#no vad
  • C. Router(config)# voice service voip Router(conf-voi-serv)#allow-connections voice-mail mod
  • D. Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
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Suggested Answer: D 🗳️

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decdca7
1 month, 1 week ago
Selected Answer: D
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html Call Forward When a call comes in on a SIP trunk and gets forwarded (CFNA / CFB / CFA), then the default behavior is for the CME to send the 302 "Moved Temporarily" SIP message to the Service Provider (SP) proxy. The user portion of the Contact Header in the 302 message might need to be translated to reflect a DID that the SP proxy can route to. The host portion of the Contact Header in the 302 message should be modified to reflect the Address of Record (AOR) using the host-registrar CLI under sip-ua and the b2bua CLI under the VoIP dial peer going to the CUE. Some SIP proxies might not support this. If so, then you need to add this: Router(config)#voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
upvoted 1 times
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jatoja83
1 month, 3 weeks ago
Selected Answer: B
Explanation: The issue you're describing suggests that inbound calls from the PSTN SIP trunk are not being routed correctly to voicemail and instead receive a busy signal after 20 seconds. This often happens because the Voice Activity Detection (VAD) feature, which can cause interruptions in call routing, may be enabled. Disabling VAD is a typical fix in such scenarios. VAD (Voice Activity Detection): This feature can cause issues in call routing if it incorrectly detects silence and prematurely ends the call. Disabling VAD ensures that the call stays active, allowing it to be routed correctly to voicemail.
upvoted 1 times
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rsl4u
9 months, 2 weeks ago
Reference URL does not exist anymore: Other URL with explaination: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html (Section: Call Forward)
upvoted 1 times
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john_doe_9999
10 months ago
Selected Answer: D
I'd say D
upvoted 1 times
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Mert_kerna
2 years, 5 months ago
Selected Answer: D
no supplementary-service sip moved temporarily is to remove the default behavior where CME sends the 302 moved temporarily SIP message to the Service Provider.
upvoted 2 times
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AgshinA
3 years, 7 months ago
it should be B. VAD is enabled by default. Because there is no comfort noise during periods of silence, the call may seem to be disconnected. You may prefer to set no vad on the SIP phone pool.
upvoted 2 times
AgshinA
3 years, 7 months ago
Actually it is D. https://community.cisco.com/t5/ip-telephony-and-phones/help-please-callers-get-busy-signal-instead-of-aim-cue-voicemail/td-p/1911590
upvoted 12 times
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Grebec94
4 years, 2 months ago
A voice activity detection (VAD) must be switched off and a codec must be designated.
upvoted 4 times
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