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Cisco 300-815 Exam - Topic 8 Question 90 Discussion

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?
D) Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
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

Cisco 300-815 Exam - Topic 8 Question 90 Discussion

Actual exam question for Cisco's 300-815 exam
Question #: 90
Topic #: 8
[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?

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Suggested Answer: D

by Garry at Jun 22, 2024, 06:39 AM

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Aretha
9 months ago
20 seconds of ringing then busy? That’s odd, something feels off here.
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Dallas
9 months ago
I’ve seen this problem before, B worked for me!
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Antione
10 months ago
Wait, are we sure it’s not option D? That sounds plausible too.
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Aleta
10 months ago
Definitely agree with B, it makes the most sense!
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Glory
10 months ago
I think option B is the right fix for this issue.
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Carey
10 months ago
I thought allowing connections to voicemail was important, so maybe option C is the correct answer?
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Luis
11 months ago
I feel like option D might be related to SIP signaling issues, but I can't recall the exact details.
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Glory
11 months ago
I think we practiced a similar question where the busy signal was caused by a misconfiguration in the voice service settings.
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Rueben
11 months ago
I remember something about configuring dial peers, but I'm not sure if "no vad" is the right fix for this issue.
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Mari
11 months ago
I've seen this type of problem before. I'm pretty confident option C is the solution - allowing connections to the voice-mail module should route the calls properly.
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Sheridan
11 months ago
Hmm, the question mentions a SIP trunk, so I'll pay close attention to the SIP-related commands. I think option D might be the right fix, but I'll double-check the other choices too.
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Katheryn
11 months ago
Okay, let me think this through. It sounds like the issue is with the voicemail not picking up the call. I'll focus on the voice service and dial-peer settings.
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Celestina
11 months ago
This seems like a tricky one. I'll need to carefully review the configuration steps to make sure I understand what's going on.
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Essie
11 months ago
Hmm, this is a tricky one. I'm a bit confused about the differences between the answer choices. I'll need to re-read the question and options closely to make sure I understand the nuances.
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Harrison
11 months ago
I'm pretty confident the answer is B. Arrays in JSON can hold ordered collections of values, which is exactly how Python tuples work. The other options like objects and dicts don't seem as relevant here.
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Youlanda
11 months ago
Hmm, I'm a bit unsure about this one. I'll need to think through the different options carefully to determine the best approach.
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Brice
11 months ago
Okay, I think I've got this. Based on the information provided, I'd say the answer is b and d. Testing for increased network traffic and ensuring no high priority defects are outstanding seem like key parts of a good test policy.
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Marylin
11 months ago
I'm pretty sure the answer is B, since we need to update the deployment without any resource downtime.
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Blythe
1 year ago
I'm leaning towards option A. Allowing H.323 connections might be the solution, even though it's not directly related to SIP. Worth a try, right?
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Vanna
1 year ago
Haha, this question is a real head-scratcher! I'm going to take a wild guess and go with option B. Disabling VAD might do the trick, or at least make the call sound more entertaining.
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Jospeh
1 year ago
User 3: I agree with Brynn, option B seems like the most logical solution here.
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Brynn
1 year ago
User 2: I'm not so sure about that. Maybe option C would be a better choice.
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Miss
1 year ago
User 1: I think option B is the way to go. Disabling VAD could solve the issue.
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Joesph
1 year ago
Option C looks good to me. Allowing connections to the voice-mail module should route the calls to Cisco Unity Express instead of the busy signal.
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Nichelle
1 year ago
C) Router(config)# voice service voip Router(conf-voi-serv)#allow-connections voice-mail mod
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Casie
1 year ago
B) Router(config)#dial-peer voice 2 voip Router(config-dial-peer)#no vad
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Shaquana
1 year ago
A) Router(config)# voice service voip Router(conf-voi-serv)#allow-connections h323 to h323
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An
1 year ago
I'm not sure, but I think B could also be a possible solution.
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Fabiola
1 year ago
I disagree, I believe the correct answer is C.
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Dulce
1 year ago
I think option D is the correct answer. Disabling the supplementary service 'sip moved-temporarily' should fix the issue with the PSTN SIP trunk calls not going to voicemail.
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Leatha
1 year ago
Option D seems like the right solution.
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Alease
1 year ago
I would go with option D as well.
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Gene
1 year ago
I believe option D is the best choice.
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Erinn
1 year ago
I think option D is the correct answer.
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Stephen
1 year ago
I think the answer is A.
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