Implementing Cisco Collaboration Core Technologies (CLCOR) practice questions

1. Question 1 of 25

   Select 2

   Which two actions must be taken to provision a new device using the self-provisioning? (Choose two.)

   • AEnable the self-provisioning IVR in the Cisco UCM.
   • BImport the user profile to the corporate LDAP directory.
   • CLink the appropriate universal device template to the user profile.
   • DEnsure the user has a directory URI and a primary extension.
   • ELink the appropriate service profile to the provisioning template.

   Show answer

   Correct answer: A, C

   How others answered

   • AC100%

   Explanation

   It should be A and C by common sense, you do not need an URI. Self-Provisioning of a phone to a user needs: * an End user with a primary extension and a Feature Group Template * the Feature Group Template needs a User Profile * the User Profile needs a Universal Line and Universal Device Template finally to activate it you need a CTI route point and a configured IVR service to actually register it. HOWEVER, it looks like you dont actually NEED the IVR to technically fullfil the goal of self-registration, as users can use the URL based registration. but i am not 100% sure if you wont need the IVR anyways, even if you arent using it - the guide is not 100% clear on that https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/214228-configure-self-provisioning-feature-on-c.html

   Adapted from community discussion by [Removed] · 2022-12-30 · 6 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 1 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. [

      [Removed]

      2022-12-30

      Selected A, C6 upvotes

      It should be A and C by common sense, you do not need an URI. Self-Provisioning of a phone to a user needs: * an End user with a primary extension and a Feature Group Template * the Feature Group Template needs a User Profile * the User Profile needs a Universal Line and Universal Device Template finally to activate it you need a CTI route point and a configured IVR service to actually register it. HOWEVER, it looks like you dont actually NEED the IVR to technically fullfil the goal of self-registration, as users can use the URL based registration. but i am not 100% sure if you wont need the IVR anyways, even if you arent using it - the guide is not 100% clear on that https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/214228-configure-self-provisioning-feature-on-c.html

   2. D

      DaKenjee

      2023-05-24

      Selected A, C5 upvotes

      URL Based Explanation by FlyingThunder is another option, but Guide below, explains other method with IVR, which is mentioned here and suitable. Problem here is that Answer A,C and D suits But when i assume question focus on activating self-provisioning, Answer D get skipt by Step 5: Peer Guide (Answer A) Activate Services for Self-Provisioning Step 4 Under CTI Services, check Self Provisioning IVR. (Answer C) Enable Autoregistration for Self-Provisioning Step 3 Select the Universal Device Template that you want to be applied to provisioned phones. Step 5 Use the Starting Directory Number and Ending Directory Number fields to enter a range of directory numbers to apply to provisioned phones. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1/sysConfig/11_5_1_SU1/cucm_b_system-configuration-guide-1151su1/cucm_b_system-configuration-guide-1151su1_chapter_01001000.html

   3. P

      pasangawa

      2024-07-10

      Selected A, C1 upvote

      https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/12_0_1/systemConfig/cucm_b_system-configuration-guide-1201/cucm_b_system-configuration-guide-1201_chapter_01001000.html#CUCM_TP_S5DD8938_00 "Your end users must be associated to a user profile or feature group template that includes a universal line template, universal device template." " In Cisco Unified Serviceability, activate the Self-Provisioning IVR and CTI Manager services."

   4. G

      G0y0

      2025-03-11

      1 upvote

      whether if method URL or method IVR, should you need an user profile. Both methods are different and independent

2. Question 2 of 25

   An administrator is trying to change the default LINECODE for a voice ISDN T1 PRI. Which command makes this change?

   • Alinecode esf
   • Blinecode ami
   • Clinecode hdb3
   • Dlinecode b8zs

   Show answer

   Correct answer: D

   How others answered

   • D57%
   • B43%

   Only 57% agreed on this answer — treat it as contested rather than settled.

   Explanation

   D is correct. Question is changing from default (which is AMI) for T1 line. So the correct answer is b8zs. hdb3 is for E1. https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/12_4t/vc_12_4t_book/vc_mgcp_t1cas.html

   Adapted from community discussion by Brant · 2022-03-07 · 9 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 2 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. B

      Brant

      2022-03-07

      9 upvotes

      D is correct. Question is changing from default (which is AMI) for T1 line. So the correct answer is b8zs. hdb3 is for E1. https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/12_4t/vc_12_4t_book/vc_mgcp_t1cas.html

   2. S

      somedudebob

      2021-10-28

      5 upvotes

      However Looking at the CLCOR book (PAGE 535) it states that AMI is default for T1 lines, so I would select option D b8zs as being the command to CHANGE the default as it is the only other valid option. I believe the first link is referring to T1 and data rather than voice, However I am treating the book, and Jason Ball as being correct.

   3. S

      Simpax_100

      2021-01-27

      4 upvotes

      Cisco CLCOR say: Configure T1 controller 1/0/0 with the cablelength, framing, linecode, and clock source: SJC-GW(config)#controller t1 0/1/0 SJC-GW(config-controller)#cablelength short 110ft SJC-GW(config-controller)#framing esf SJC-GW(config-controller)#linecode b8zs SJC-GW(config-controller)#clock source network Note The framing esf and linecode b8zs commands are defaults.

   4. A

      Alan100

      2023-09-23

      Selected D2 upvotes

      Answer is D. Default for T1s is AMI (Alternate Mark Inversion), Default for E1s is hdb3 (High Density Bipolar 3). b8zs (Bipolar with eight-zero substitution) is a modified AMI for North American States. https://en.wikipedia.org/wiki/Integrated_Services_Digital_Network. According to Cisco, AMI is already the default for T1s so to change it, the only other option is b8zs. https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/12_4t/vc_12_4t_book/vc_mgcp_t1cas.html

3. Question 3 of 25

   

   Refer to the exhibit. What causes of the PRI issue?

   • AThe cable is unplugged.
   • BThe clock source is incorrect.
   • CThe controller shut down.
   • DThe framing is configured incorrectly.

   Show answer

   Correct answer: D

   How others answered

   • D69%
   • B31%

   Only 69% agreed on this answer — treat it as contested rather than settled.

   Explanation

   Yes, A and C, are wrong, you wont have layer 1 up if they are true. and B clock problems, cant turn down the layer 2 service.

   Adapted from community discussion by santiagof · 2022-11-06 · 5 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 3 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. G

      G0y0

      2025-03-10

      Selected D1 upvote

      A. and C. do not have sense. B, is talking about a possibility of a intermittent service, it could be swinging between up and down. These errors can be viewed in the show controllers T1, however, the Layer 2 can be viewed in state MULTIPLE_FRAME_ESTABLISHED even though these slips errors. D. could be the best answer, simply on side can not understand the frames from the other side, mean while while one side is sending SABME's like crazy and not receiving anything from the other side that could be solved with a correct configuration. I do not know if the problem is the framing or a bad configuration in the Serial interface (isdn protocol-emulate command), because it is stupid to be guessing without without having debug isdn q921and show controller t1 outputs, however, the only reason about the problem is a configuration issue, and just the answer D shows us "configured incorrectly". Something is configured incorrectly, without a doubt.

   2. S

      santiagof

      2022-11-06

      5 upvotes

      Yes, A and C, are wrong, you wont have layer 1 up if they are true. and B clock problems, cant turn down the layer 2 service.

   3. J

      JoeC716

      2024-05-06

      Selected D2 upvotes

      Read MeowthL's comment and go to the link, in there read about the TEI_ASSIGNED state in the Layer 2 status and it talks about BAD FRAMES

   4. G

      G0y0

      2025-03-10

      1 upvote

      Furthermore, the show isdn status command displays a summary of the current status. Therefore, Layer 2 can bounce up and down even though it indicates a MULTIPLE_FRAME_ESTABLISHED state. It is needed to use the debug isdn q921 command to ensure that Layer 2 is stable. This situation can be leveraged by a incorrect clock, however, as we do not have debug isdn q921, we are forced to assume that clock is correct and no slip erros. Please refer to (it is amazing this document): Troubleshoot T1 PPRI https://www.cisco.com/c/en/us/support/docs/wan/t1-e1-t3-e3/8131-T1-pri.pdf

4. Question 4 of 25

   Which command is used in Cisco IOS XE TDM gateway to configure the voice T1/E1 controller to provide clocking?

   • Aclock source line
   • Bclock source external
   • Cclock source internal
   • Dclock source network

   Show answer

   Correct answer: D

   How others answered

   • D77%
   • C23%

   Explanation

   https://www.cisco.com/c/en/us/td/docs/routers/access/interfaces/NIM/software/configuration/guide/4gen-t1-e1-nim-guide.html To provide clocking to the line use the command clock source network . The command clock source internal is applicable to data T1/E1 and is not used for T1/E1 voice.

   Adapted from community discussion by Ahochau · 2023-12-01 · 2 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 4 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. O

      Omitted

      2022-11-05

      Selected D7 upvotes

      Clock source network - should be configured voice. If the t1 is data only then you would do clock source internal

   2. A

      Ahochau

      2023-12-01

      Selected D2 upvotes

      https://www.cisco.com/c/en/us/td/docs/routers/access/interfaces/NIM/software/configuration/guide/4gen-t1-e1-nim-guide.html To provide clocking to the line use the command clock source network . The command clock source internal is applicable to data T1/E1 and is not used for T1/E1 voice.

   3. L

      leodev

      2022-12-07

      3 upvotes

      Its D. The command clock source internal is applicable to data T1/E1 and is not used for T1/E1 voice.

   4. S

      sneff91

      2024-09-04

      1 upvote

      Not sure where you got that from. I literally see "clock source network" on my IOS XE router running 17.06.05: R1(config-controller)#clock source ? internal Internal Clock line Line Recovered Clock network Internal TDMSwitch Source R1(config-controller)#end R1#sh vers Cisco IOS XE Software, Version 17.06.05 Cisco IOS Software [Bengaluru], c8000be Software (X86_64_LINUX_IOSD-UNIVERSALK9-M), Version 17.6.5, RELEASE SOFTWARE (fc2)

5. Question 5 of 25

   Which command in the MGCP gateway configuration defines the secondary Cisco Unified Communications Manager server?

   • Amgcpapp
   • Bccm-manager fallback-mgcp
   • Cmgcp call-agent
   • Dccm-manager redundant-host

   Show answer

   Correct answer: D

   How others answered

   • D100%

   Explanation

   D https://www.cisco.com/c/en/us/td/docs/routers/access/vgd1t3/rel1_0/software/configuration/guide/VGD_mgcp.html ccm-manager fallback-mgcp - Enables the MGCP fallback feature ccm-manager redundant-host {ip-address | DNS-name} [ip-address | DNS-name] - Identifies up to two backup Cisco Unified Communications Manager servers.

   Adapted from community discussion by Puh · 2022-02-14 · 2 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 5 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. V

      Vincentius

      2021-05-25

      26 upvotes

      Should be D: ccm-manager redundant-host https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/12_4t/vc_12_4t_book/vc_ucm_mgcp_gw.html

   2. D

      Dread

      2022-03-04

      8 upvotes

      Guys has anyone passed recently after using these materials. They seem to be wrong most of them!?

   3. R

      rishik

      2021-07-17

      4 upvotes

      ccm-manager mgcp ccm-manager music-on-hold ccm-manager config server X.X.X.X Primary server ccm-manager redundant-host X.X.X.X Secondary server ccm-manager config ! mgcp mgcp call-agent X.X.X.X 2427 service-type mgcp version 0.1

   4. B

      BarryR

      2021-07-01

      4 upvotes

      Answer is D...ccm-manager redundant-host {ip-address | DNS-name} [ip-address | DNS-name]

6. Question 6 of 25

   Refer to the exhibit.

   

   An administrator configures a secure SIP trunk on Cisco UCM. Which value is needed in the Secure Certificate Subject or Subject Alternate Name field to accomplish this task?

   • Athe common name of the remote device certificates
   • Bthe fully qualified domain name of all Cisco UCM nodes that run the CallManager service
   • Cthe common name of the Cisco UCM CallManager certificate
   • Dthe fully qualified domain name of the remote device that is configured on the SIP trunk

   Show answer

   Correct answer: D

   How others answered

   • D48%
   • A45%

   Only 48% agreed on this answer — treat it as contested rather than settled.

   Explanation

   Have a Look at this Link (https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200180-Configure-SIP-TLS-Trunk-on-the-Communica.html ). Section: Step 8. Create SIP Trunk Security Profiles. Here you can See that it is the CN of the remote Host/devices. At the screenshots below you can See that a CN is not always a FQDN. In this example it is CUCM10.

   Adapted from community discussion by SergeantDuty · 2023-07-26 · 3 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 6 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. S

      Slushed

      2022-05-02

      Selected D12 upvotes

      The correct answer is D. https://video.cisco.com/video/6196744148001

   2. A

      azizkasmir

      2022-12-12

      Selected D4 upvotes

      https://www.google.com/url?sa=t&rct=j&q=&esrc=s&source=web&cd=&cad=rja&uact=8&ved=2ahUKEwik1rGj9PT7AhVTH7cAHUcHBFgQFnoECBMQAw&url=https%3A%2F%2Fwww.youtube.com%2Fwatch%3Fv%3DT1IEHMCQPb0&usg=AOvVaw0_ghphBkOhI3Ob2_-mXaZI

   3. S

      SergeantDuty

      2023-07-26

      3 upvotes

      Have a Look at this Link (https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200180-Configure-SIP-TLS-Trunk-on-the-Communica.html ). Section: Step 8. Create SIP Trunk Security Profiles. Here you can See that it is the CN of the remote Host/devices. At the screenshots below you can See that a CN is not always a FQDN. In this example it is CUCM10.

   4. G

      G0y0

      2025-02-27

      1 upvote

      In the reference you provide, you are talking about a CUC certificate, which does match the FQDN, but the question does not tell you what is on the other side, whether it is a UNITY, an IM&T, a CUBE, a Gateway, an Expressway-C, or a cluster, or an Oracle SBC, etc. etc. etc. FQDN is not the same as CN. If the question said that the remote device is a CUC, D would be correct. But assuming that it is the FQDN of a remote device that we do not know, the first thing we have to see is the CN and in this case it is precisely the one that is not used, therefore it is A. It is recommended that if you are going to cite a source, it is valued. But please, interpret and read the source correctly before advertising it.

7. Question 7 of 25

   A collaboration engineer must configure Cisco Unified Border Element to support up to five concurrent outbound calls across an Ethernet link with a bandwidth of 160 kb to the Internet Telephony Service Provider. Which set of commands allows the engineer to complete the task without compromising voice quality?

   • Adial-peer voice 1 voip translation-profile outgoing Strip9 max-conn 5 destination-pattern 91[2-9]..[2-9]ג€¦ג€¦$ session protocol sipv2 session target ipv4:142.45.10.1 dtmf-relay rtp-nte sip-notify sip-kpml
   • Bdial-peer voice 1 voip translation-profile outgoing Strip9 max-conn 5 destination-pattern 91[2-9]..[2-9]ג€¦ג€¦$ session protocol sipv2 session target ipv4:142.45.10.1 dtmf-relay rtp-nte sip-notify sip-kpml codec ilbc mode 20
   • Cdial-peer voice 1 voip translation-profile outgoing Strip9 max-conn 5 destination-pattern 91[2-9]..[2-9]ג€¦ג€¦$ session protocol sipv2 session target ipv4:142.45.10.1 dtmf-relay rtp-nte sip-notify sip-kpml codec aacld
   • Ddial-peer voice 1 voip translation-profile outgoing Strip9 max-conn 5 destination-pattern 91[2-9]..[2-9]ג€¦ג€¦$ session protocol sipv2 session target ipv4:142.45.10.1 dtmf-relay rtp-nte sip-notify sip-kpml codec mp4a-latm

   Show answer

   Correct answer: A

   How others answered

   • A82%

   Explanation

   but iblc 20 it's 38,4 kbps => x5 = 192 kps => more than 160 kps correct answer is not "A"? Becouse without codec value in the dial-peer, by default it's use G729, and G729 is 31.2 kps => x5 = 156 kps

   Adapted from community discussion by spag22500 · 2022-09-01 · 11 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 7 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. S

      spag22500

      2022-09-01

      11 upvotes

      but iblc 20 it's 38,4 kbps => x5 = 192 kps => more than 160 kps correct answer is not "A"? Becouse without codec value in the dial-peer, by default it's use G729, and G729 is 31.2 kps => x5 = 156 kps

   2. D

      DaKenjee

      2024-11-24

      Selected A5 upvotes

      As Omitted explains: default for voip Dial-peer is G.729, which suits all other dial-peers have same value, but use different codecs, which consume more bandwith

   3. C

      CollabGuy

      2022-11-03

      5 upvotes

      Damn. I chose iLBC instinctively, but I also think you are right. g729r8, 20-byte payload is configured by default and total would be 156kbps. Tricky question...

   4. F

      F3rnando

      2023-08-24

      4 upvotes

      https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html G.729: 31.2×5= 156kb ilbc: 38.4×5= 192kb so its A for me

8. Question 8 of 25

   Calls are being delivered to the end user in the globalized format. Where does an engineer configure the calling number into a localized format?

   • Aroute pattern
   • Bservice parameters
   • CIP phone
   • Dgateway

   Show answer

   Correct answer: C

   How others answered

   • C50%
   • D50%

   Only 50% agreed on this answer — treat it as contested rather than settled.

   Explanation

   The answer is C -Ip Phone - For the inbound call flow, globalization is done at ingress in the sip trunk/ gateway and localization at egress in the phone level with a calling party transformation. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11/dialplan.html#16685

   Adapted from community discussion by norealchaos · 2020-08-16 · 12 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 8 discussion

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   1. N

      norealchaos

      2020-08-16

      12 upvotes

      The answer is C -Ip Phone - For the inbound call flow, globalization is done at ingress in the sip trunk/ gateway and localization at egress in the phone level with a calling party transformation. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11/dialplan.html#16685

   2. K

      Komy

      2021-07-01

      5 upvotes

      I would say the answer is "Gateway". Here is my reasoning: - Localization can be configured on Gateway and/or Device pool and/or phone level. - However, In SRND 12, Under (Phone calling party number Localization) , the below statement shows that Cisco recommends placing the configuration in device pool and not on the phone: "The Calling Party Transformation Pattern is placed in a partition included in the destination phone's Calling Party Transformation Pattern CSS, configured at the device-pool level." And because we do not have "device pool" as an available answer, i would go with "Gateway" because you can't just go to each and every single phone to apply config

   3. D

      DaKenjee

      2022-11-24

      Selected C2 upvotes

      Answer is C - Komy, explained it, with SRND12 I searched and found a similar answer in SRND route pattern is wrong, directs from CUCM away service parameters, is possible -> service parameters to globalize calling party numbers on all devices clusterwide IP phone, is possible with calling party transformation CSS gateway, is possible by varios manipulations via CLI i stick with Answer C, because gateway is ealiest ingrees in this callflow: ITSP > Gateway > Callmanager > Phone SRND: Thus, the guiding principle is: Accept localized forms upon call ingress, and globalize them; route the call based on the globalized form; and localize the call to comply to the form required by the destination. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/dialplan.html?bookSearch=true#48778

   4. M

      Mert_kerna

      2023-01-27

      Selected C3 upvotes

      The answer is C, because the calling party transformation CSS can only be applied to an IP phone or a device pool. Device pool is not a given answer, so, the answer is IP Phone. The associated transformation pattern, however, can be applied directly on the device pool or the gateway. The question is asking where the configuration can be applied. The final configuration (CSS) "Configuration" can only be applied to the Device Pool or an IP Phone. Not the gateway. -

9. Question 9 of 25

   How does an administrator make a Cisco IP phone display the last 10 digits of the calling number when the call is in the connected state, and also display the calling number in the E.164 format within call history on the phone?

   • AConfigure a translation pattern that has a Calling Party Transform Mask of XXXXXXXXXX.
   • BOn the inbound SIP trunk, change Significant Digits to 10.
   • CChange the service parameter Apply Transformations On Remote Number to True.
   • DConfigure a calling party transformation pattern that keeps only the last 10 digits.

   Show answer

   Correct answer: D

   How others answered

   • D80%
   • C20%

   Explanation

   Answer is D Translationpattern might work, but have to be triggered as called number equals directory number, not suitable sip-trung can change digits on calling number Calling Party Transformations on IP Phones Calling Party Transformation Patterns allow the system to adapt the calling party numbers to different formats. The most typical use is to adapt from globalized to localized calling party numbers and vice versa. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11/dialplan.html

   Adapted from community discussion by DaKenjee · 2024-11-25 · 3 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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   Question 9 discussion

   These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

   1. R

      ratbat

      2022-11-17

      8 upvotes

      D is the answer and here's why: if your read scenario 2 on the below link, you will know it can only be D https://blog.collabcert.com/ucm/understanding-the-5-uses-of-calling-party-transformation-pattern-in-ucm/

   2. D

      DaKenjee

      2024-11-25

      Selected D3 upvotes

      Answer is D Translationpattern might work, but have to be triggered as called number equals directory number, not suitable sip-trung can change digits on calling number Calling Party Transformations on IP Phones Calling Party Transformation Patterns allow the system to adapt the calling party numbers to different formats. The most typical use is to adapt from globalized to localized calling party numbers and vice versa. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11/dialplan.html

   3. K

      Komy

      2023-07-02

      4 upvotes

      I will go with C: below is captured from SRND 12: "The outbound call’s calling party transformation CSS (also referred to as localization or remote number calling party transformation CSS) can also be used to localize remote connected party information. To enable this, the advanced service parameter Apply Transformations On Remote Number must be enabled."

   4. J

      jonycakes

      2023-06-10

      2 upvotes

      Answer D: On a phone in North America, Cisco Unified Communications Manager uses a calling party transformation to convert +12225551234 to 10 digits before the number displays on the phone; on a phone outside of North America, Cisco Unified Communications Manager may transform the number to only strip the + and to prefix the 00, as in 0012225551234. Source: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_01010.html

10. Question 10 of 25

    

    Refer to the exhibit. A user takes a phone from San Francisco to New York for a short reassignment. The phone was set up to use the San Francisco device pool, and device mobility is enabled on the Cisco UCM. The user makes a call that matches a route pattern in a route list that contains the Standard Local Route Group. To where does the call retreat?

    • AThe call egresses in San Francisco because the user uses device mobility and is allowed to roam while still keeping the number and resources assigned in San Francisco.
    • BThe call egresses in New York because the device automatically is assigned a New York device pool and uses the local gateway.
    • CThe call fails because device mobility is turned on, and the phone is not configured in New York. The engineer must configure which sites the device should be roaming to.
    • DThe call fails because the Standard Local Route Group is being used only if no configuration is set for the device pools.

    Show answer

    Correct answer: B

    How others answered

    • B94%

    Explanation

    Correct answer is B. The point of Device Mobility is for devices that are moved to a new location to utilize the resources local to that location.

    Adapted from community discussion by Slushed · 2023-05-02 · 10 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    Question 10 discussion

    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. S

       Slushed

       2023-05-02

       Selected B10 upvotes

       Correct answer is B. The point of Device Mobility is for devices that are moved to a new location to utilize the resources local to that location.

    2. D

       DaKenjee

       2023-11-25

       Selected B4 upvotes

       Answer B Standard local route group would use San Francisco device pool, if there is no device mobility active Device Mobility Info Configuration delivers by IPv4 Subnet a selected Device Pool ; in this case new york device pool and its local gateway in new york

    3. G

       G0y0

       2025-02-27

       2 upvotes

       If certain calls from roaming endpoints need to be routed through gateways local to the home site of the roaming phone, then routing for these calls has to be implemented through route patterns pointing to route lists that use fixed site-specific route groups instead of Standard Local Group. If this were the case, then A would be the correct answer. References: "Feature Configuration Guide for Cisco Unified Communications Manager, Release 12.5(1)", Chapter 4: Device Mobility, Table 4: Device Mobility Scenarios. "Cisco Collaboration System 12.x Solution Reference Network Designs SRND", Chapter 21: Mobile Collaboration, section: "Dial Plan Design Considerations"

    4. D

       DaKenjee

       2023-11-25

       3 upvotes

       They used pictures and explanation from SRND Example 14-3 Device Mobility https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/dialplan.html

11. Question 11 of 25

    A presence redundancy group is deployed, and an engineer initiates a manual fallback. Which statement about Cisco Server Recovery Manager is true?

    • Adisconnects all users that had been failed over, and the users must log in again
    • Bdisconnects all users that had been failed over
    • Crestarts critical services on the secondary node
    • Drestarts the Cisco Presence Engine

    Show answer

    Correct answer: A

    How others answered

    • A70%
    • B30%

    Only 70% agreed on this answer — treat it as contested rather than settled.

    Explanation

    Thinking this should be A: Manual fallback—When you initiate a manual fallback, the Cisco Server Recovery Manager restarts critical services on the primary node and disconnects all users that had been failed over. Those users must then re-login to their assigned node. Source: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide-1101/CUCM_BK_C733E983_00_cucm-system-configuration-guide-transformed_chapter_011010.html

    Adapted from community discussion by jonycakes · 2021-12-07 · 13 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    Question 11 discussion

    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. J

       jonycakes

       2021-12-07

       13 upvotes

       Thinking this should be A: Manual fallback—When you initiate a manual fallback, the Cisco Server Recovery Manager restarts critical services on the primary node and disconnects all users that had been failed over. Those users must then re-login to their assigned node. Source: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide-1101/CUCM_BK_C733E983_00_cucm-system-configuration-guide-transformed_chapter_011010.html

    2. O

       Omitted

       2022-01-26

       2 upvotes

       B is the best fit in my opinion. I don't like A because it doesn't acknowledge that users shouldn't have to relog in. "In 10.X and later...The user does not have to exit Jabber and re-log in when the change happens with Server Recovery Manager(SRM)." https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-im-presence-service/200958-IM-and-Presence-Server-High-Availability.html#anc7 It's not C bc this is only true for the primary node.. "Cisco Server Recovery Manager restarts critical services on the primary node and disconnects all users that had been failed over" https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/im_presence/configAdminGuide/11_5_1/cup0_b_config-and-admin-guide-1151su5/cup0_b_imp-system-configuration-1151su5_chapter_0100.html#task_6D63EA716B4BFA54377429B6C5D1E7C3

    3. V

       VG224

       2022-04-01

       2 upvotes

       B is correct , users doesn't have to re-login User Failover Process When a failover takes place(automatic or manual), the major point to remember is that the user account is not moved from one server to the other, but only the user session in Presence Engine is moved. In pre-10 versions of IM and Presence, the user assignment was moved from one server to the other. This user move was very expensive to server resources and added to the load that was on the server. In 10.X and later, the user stays homed on the server that they are assigned to, and the backend user session in the Presence Engine is moved from the failed node to the functional node. The user does not have to exit Jabber and re-log in when the change happens with Server Recovery Manager(SRM).

    4. G

       G0y0

       2025-03-11

       Selected A1 upvote

       After a failover, when the failed node comes online again, the clients automatically reconnect to the local IM and Presence Service node if you have configured automatic fallback. If you have not configured automatic fallback, you can manually initiate the fallback when the failed node comes online. Now, when you initiate a manual fallback, the Cisco Server Recovery Manager restarts critical services on the primary node and disconnects all users that had been failed over. Those users must then re-login to their assigned node. So, answer is A. Reference: Configure Redundancy and High Availability https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/im_presence/configAdminGuide/11_5_1/cup0_b_config-and-admin-guide-1151su5/cup0_b_imp-system-configuration-1151su5_chapter_0100.pdf

12. Question 12 of 25

    Select 2

    When a call is delivered to a gateway, the calling and called party number must be adapted to the PSTN service requirements of the trunk group. If a call is destined locally, the + sign and the explicit country code must be replaced with a national prefix. For the same city or region, the local area code must be replaced by a local prefix as applicable. Assuming that a Cisco UCM has a SIP trunk to a New York gateway (area code 917), which two combinations of solutions localize the calling and called party for a New York phone user? (Choose two.)

    • AConfigure the gateway to translate the calling number and apply it to the dial peer. Combine it with a translation profile for called numbers. ! voice translation-rule 1 rule 1 /^1917/ // rule 2 /^ [+]1917/ // ! voice translation-profile strip+1 translate calling 1 !
    • BConfigure two calling party transformation patterns: \+1917.CCCCCC, strip pre-dot, numbering type: subscriber \+!, strip pre-dot, numbering type: national
    • CConfigure two called party transformation patterns: \+1917.XXXXXXX, strip pre-dot, numbering type: subscriber \+1.!, strip pre-dot, numbering type: national
    • DConfigure the gateway to translate called numbers and apply it to the dial peer. Combine it with a translation profile for calling numbers. ! voice translation-rule 1 rule 1 /^1917/ // rule 2 /^[+]1917/ // ! voice translation-profile strip+1 translate called 1 !
    • EConfigure two calling party transformation patterns: \+1917.XXXXXXX, strip pre-dot, numbering type: subscriber \+1.!, strip pre-dot, numbering type: national

    Show answer

    Correct answer: C, E

    How others answered

    • CE64%
    • AD21%

    Only 64% agreed on this answer — treat it as contested rather than settled.

    Explanation

    Answer C+E First we have to adjust calling + called and a mixture by gateway normalization + callmanager normalization makes no sence. Answer E is working for +1917 local->subscriber and +1 -> national signaling Gateway Calling Party Number Localization: As a call is delivered to a gateway, the calling party number must be adapted to the requirements of the PSTN service provider providing the trunk group to which the gateway is connected. Calling Party Number Transformation patterns can be used to change the calling party number digit string and numbering type. They even use same wording like in question: \+1.!, strip pre-dot, numbering type: national When a call egresses to a gateway, the calling and called party transformation patterns are applied to the calling and called numbers respectively. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/dialplan.html

    Adapted from community discussion by DaKenjee · 2022-11-25 · 9 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. D

       DaKenjee

       2022-11-25

       Selected C, E9 upvotes

       Answer C+E First we have to adjust calling + called and a mixture by gateway normalization + callmanager normalization makes no sence. Answer E is working for +1917 local->subscriber and +1 -> national signaling Gateway Calling Party Number Localization: As a call is delivered to a gateway, the calling party number must be adapted to the requirements of the PSTN service provider providing the trunk group to which the gateway is connected. Calling Party Number Transformation patterns can be used to change the calling party number digit string and numbering type. They even use same wording like in question: \+1.!, strip pre-dot, numbering type: national When a call egresses to a gateway, the calling and called party transformation patterns are applied to the calling and called numbers respectively. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/dialplan.html

    2. W

       wwisp3422112

       2022-11-30

       4 upvotes

       I agree with DaKenjee: CE correct. If u want to remember the answers to the exam just think about: "XXXXXXX"

    3. O

       OSJAY

       2022-09-11

       3 upvotes

       None of the answers convince me. I would do: voice translation-rule 1 rule 1 /^1917/ // rule 2 /^[+]1/ // ! voice translation-profile strip+1 translate called 1 translate calling 1 Apply as Outbound profile to outbound DP

    4. W

       WeNt48

       2023-03-02

       3 upvotes

       Correct. SIP Trunks doesn't transmit numbering type information in signalization. Whole configuration must be completed on gateway, and that means only A and D are correct.

13. Question 13 of 25

    A Cisco Unity Connection Administrator must set a voice mailbox so that it can be accessed from a secondary device. Which configuration on the voice mailbox makes this change?

    • AAttempt Forward routing rule
    • BAlternate Extensions
    • CAlternate Names
    • DMobile User

    Show answer

    Correct answer: B

    How others answered

    • B100%

    Explanation

    Alt. Ext is correct. lternate Extensions In addition to the primary extension for each user, you can set up alternate extensions. Alternate extensions can be used for various reasons, such as handling multiple line appearances on user phones. Alternate extensions can also make calling Unity Connection from an alternate device, such as a mobile phone or a phone at another work site more convenient. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx/10xcucsagappa.html#92809

    Adapted from community discussion by Abdis · 2022-09-09 · 4 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. A

       Abdis

       2022-09-09

       4 upvotes

       Alt. Ext is correct. lternate Extensions In addition to the primary extension for each user, you can set up alternate extensions. Alternate extensions can be used for various reasons, such as handling multiple line appearances on user phones. Alternate extensions can also make calling Unity Connection from an alternate device, such as a mobile phone or a phone at another work site more convenient. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx/10xcucsagappa.html#92809

    2. J

       j3344567

       2024-06-12

       2 upvotes

       B. Alternate Extensions The alternate extension is not required but increases productivity and improves the user experience. You can set up alternate extensions, such as the mobile and home phone numbers, so the user can dial the CUC pilot number from these devices and use the personal login, where only the PIN needs to be entered. If a nondefined extension calls in to the CUC system, the standard opening greeting is played by default. Figure 24-7 illustrates the main parameters that need to be configured when creating a new user in CUC manually.

    3. J

       jherlitzke

       2022-07-24

       2 upvotes

       Has anyone confirmed which answer Cisco is looking for. Alternate Ext seems right, but why do all sample question sites state Attempt forward routing rule?

    4. W

       WilliamC

       2023-07-17

       1 upvote

       I found this: Configure the Call Routing Rule This method re-routes any calls that are forwarded to the Cisco Unity system from the secondary phone (extension 6007) to the extension number of the primary phone (6001). https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unity/41600-2-ip-phones-1-mbox.html

14. Question 14 of 25

    Which Cisco Unified Communications Manager configuration is required for SIP MWI integrations?

    • ASelect ג€Redirecting Diversion Header Delivery - Inboundג€ on the SIP trunk.
    • BEnable ג€Accept presence subscriptionג€ on the SIP Trunk Security Profile.
    • CEnable ג€Accept unsolicited notificationג€ on the SIP Trunk Security Profile.
    • DSelect ג€Redirecting Diversion Header Delivery - Outboundג€ on the SIP trunk.

    Show answer

    Correct answer: C

    How others answered

    • C100%

    Explanation

    https://community.cisco.com/t5/collaboration-voice-and-video/understanding-troubleshooting-mwi-on-unity-connection/ta-p/3162948 C Enable Accept unsolicited notification on the SIP Trunk Security Profile. In case of SIP integration, CUC sends SIP notify messages to the phone system to turn the MWI on or OFF so “Accept unsolicited notification” should be checked under SIP security profile assigned to the trunk

    Adapted from community discussion by F3rnando · 2023-08-25 · 8 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. F

       F3rnando

       2023-08-25

       8 upvotes

       https://community.cisco.com/t5/collaboration-voice-and-video/understanding-troubleshooting-mwi-on-unity-connection/ta-p/3162948 C Enable Accept unsolicited notification on the SIP Trunk Security Profile. In case of SIP integration, CUC sends SIP notify messages to the phone system to turn the MWI on or OFF so “Accept unsolicited notification” should be checked under SIP security profile assigned to the trunk

    2. S

       sieteunoseis

       2023-08-04

       2 upvotes

       In case of SIP integration, CUC sends SIP notify messages to the phone system to turn the MWI on or OFF so “Accept unsolicited notification” should be checked under SIP profile assigned to the trunk. https://community.cisco.com/t5/collaboration-voice-and-video/understanding-troubleshooting-mwi-on-unity-connection/ta-p/3162948

    3. L

       landrey

       2023-06-15

       1 upvote

       According https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/integration/guide/cucm_sip/b_cucintcucmsip.pdf Select "Redirecting Diversion Header Delivery - Inbound" on the SIP trunk -> valid answer Select "Redirecting Diversion Header Delivery - Outbound" on the SIP trunk -> valid answer Enable "Accept unsolicited notification" on the SIP Trunk Security Profile. -> valid answer Anybody has any clues on this question?

    4. A

       AAMM

       2024-05-27

       1 upvote

       C https://community.cisco.com/t5/collaboration-voice-and-video/understanding-troubleshooting-mwi-on-unity-connection/ta-p/3162948#:~:text=In%20case%20of%20SIP%20integration%2C%20CUC%20sends%20SIP%20notify%20messages%20to%20the%20phone%20system%20to%20turn%20the%20MWI%20on%20or%20OFF%20so%20%E2%80%9CAccept%20unsolicited%20notification%E2%80%9D%20should%20be%20checked%20under%20SIP%20profile%20assigned%20to%20the%20trunk.

15. Question 15 of 25

    Select 2

    Which protocols does Cisco IM and Presence use to authenticate Jabber?

    • AXMPP
    • BSOAP
    • CTCP
    • DLDAP
    • EQBE

    Show answer

    Correct answer: A, B

    How others answered

    • AB100%

    Explanation

    Correct answer:AB LDAP is not a requirement. the End Users could be also Local users. IMP Login has two stages namely SOAP Login which deals with User authentication and then XMPP Login which deals with XMPP session creation and Stream Management. more details: https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber/212471-jabber-login.html#anc5

    Adapted from community discussion by mzougari · 2022-02-26 · 10 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. M

       mzougari

       2022-02-26

       10 upvotes

       Correct answer:AB LDAP is not a requirement. the End Users could be also Local users. IMP Login has two stages namely SOAP Login which deals with User authentication and then XMPP Login which deals with XMPP session creation and Stream Management. more details: https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber/212471-jabber-login.html#anc5

    2. G

       gottalearnsometime

       2022-03-10

       8 upvotes

       A and B: "IMP Login has two stages namely SOAP Login which deals with User authentication and then XMPP Login which deals with XMPP session creation and Stream Management."

    3. T

       Totor27

       2024-08-18

       2 upvotes

       You're right, the question asks what protocol is used between IMP and Jabber (not CUCM and Jabber). LDAP can be a connection protocol used, but not by the IMP The correct answer is SOAP and XMPP: https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber/212471-jabber-login.html "Jabber user Authentication with CUCM to retrieve the Service profile details which contains IMP server, Voicemail, Conferencing, CTI server details and also Device Configuration file for Phone services. Once the CUCM Log in is successful, Jabber logs in to the IM&P server to authenticate and retrieve the contact list and other IM services. The IM&P Log in has two stages namely: SOAP Log in which deals with user authentication, and then XMPP Log in which deals with XMPP session creation and Stream Management."

    4. D

       DaKenjee

       2024-05-25

       Selected A, B3 upvotes

       The IM&P Log in has two stages namely: SOAP Log in which deals with user authentication, and then XMPP Log in which deals with XMPP session creation and Stream Management. https://www.cisco.com/c/en/us/support/docs/unified-communications/jabber/212471-jabber-login.html

16. Question 16 of 25

    Regarding SIP integrations with Cisco Unified Communications Manager, if the Cisco Unity Connection is configured to listen for incoming IPv4 and IPv6 traffic, how should the addressing mode be set up in the Cisco Unity Connection?

    • ASet up is not required.
    • BSet up for each group to use IPv4 and IPv6.
    • CSet up media ports for each port group to use IPv4.
    • DSet up IPv4 and IPv6 in Cisco Unified CM.

    Show answer

    Correct answer: A

    How others answered

    • A100%

    Explanation

    Correct answer is B...On the Edit General Configuration page, for IP Addressing Mode, select the option from the list to control where Connection listens for incoming traffic: • IPv4 • IPv6 • IPv4 and IPv6

    Adapted from community discussion by BarryR · 2020-07-01 · 10 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. B

       BarryR

       2020-07-01

       10 upvotes

       Correct answer is B...On the Edit General Configuration page, for IP Addressing Mode, select the option from the list to control where Connection listens for incoming traffic: • IPv4 • IPv6 • IPv4 and IPv6

    2. O

       obie2930

       2020-08-20

       6 upvotes

       Correct answer is B based on the below description from the design guide for CUC: For SIP integrations with Cisco Unified CM, if Unity Connection is configured to listen for incoming IPv4 and IPv6 traffic, you can configure the addressing mode that Unity Connection uses for call control signaling for each port group to use either IPv4 or IPv6. (This mode is also used when connecting to a TFTP server.) In addition, you can configure the addressing mode that Unity Connection uses for media for each port group to use either IPv4 or IPv6.

    3. C

       Calas

       2021-02-22

       3 upvotes

       Second part of my reasoning is this: If Unity Connection uses IPv6 or dual-mode (IPv4 and IPv6) to communicate with Cisco Unified CM, do the following subtasks: 1. Enable IPv6 on the Unity Connection server. 2. In Cisco Unity Connection Administration, on the System Settings > General Configuration page, select an option for IP Addressing Mode to control where Unity Connection listens for incoming traffic. You can select IPv4, IPv6, or IPv4 and IPv6. The setting defaults to IPv4. The default settings are IPv4, CUCM will communicate to Unity. Answer A still applies since configurations are not really needed unless you want to use IPv4 and IPv6 in tandem but this is not a port group setting and that disqualify answer B and C. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/integration/guide/cucm_sip/b_cucintcucmsip.pdf

    4. S

       sneff91

       2024-03-09

       Selected A2 upvotes

       "IP Addressing Mode" is the name of the parameter in CUC, under System Settings -> General Configuration. If you go to the Help -> This Page menu, it says this about that parameter: "When Unity Connection is configured for IPv6, select the option from the list to control where Unity Connection listens for incoming traffic: IPv4 IPv6 IPv4 and IPv6 Default Setting: IPv4" Per the above, it DOESN'T listen for incoming traffic for IPv6 by default, but the question says "if the Cisco Unity Connection is configured to listen for incoming IPv4 and IPv6 traffic", so to me, that implies that it has ALREADY correctly been set to "IPv4 and IPv6". If it has already correctly been set, then no additional setup is required, so A. There are no IPv4/IPv6 settings in the port or port group settings, so B and C are not a thing, and D refers to a CUCM config, not Unity.

17. Question 17 of 25

    Which Cisco Unity Connection handler plays a greeting that announces the option to dial a user extension by default?

    • Athe Operator call handler
    • Bthe Goodbye call handler
    • Cthe Directory handler
    • Dthe Interview handler

    Show answer

    Correct answer: C

    How others answered

    • C46%
    • B31%
    • A23%

    Only 46% agreed on this answer — treat it as contested rather than settled.

    Explanation

    from provided link: https://www.cisco.com/en/US/docs/voice_ip_comm/unity/3x/administration/guide/313/SAG_0200.html you can see: "By default, the Operator call handler allows callers to dial the appropriate extension to reach the Sign-in conversation," So I think Answer A

    Adapted from community discussion by Kabimas66 · 2024-03-17 · 2 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. G

       G0y0

       2025-03-10

       Selected C1 upvote

       First, the question targets to the default behavior: A. and B. are incorrect. By default, the Operator call handler enables the caller to dial “0” to contact the user defined as an operator. By default, there is not an extension configured for this transfer, meaning the operator is not be available. At this point, the Operator call handler is configured to take a message. After the caller has recorded a message for the operator, the user is directed to the Goodbye call handler, which simply says goodbye and disconnects the call. D. is incorrect. Interview handlers, are intended primarily for recording the caller’s messages and delivering these messages to a single user or group of users. So, C. is correct since by Default, if there is caller exists, automatically enter the Opening Greeting Handler, and this one by default allow the user to enter the extension. So correct answer is C. Cisco Unity Connection - David Schulz Chapter 8: Understanding Call Handlers and System Features

    2. G

       gorzad

       2022-12-29

       2 upvotes

       The Goodbye call handler allows the caller to sign in, restart the opening greeting, reach the operator, or dial an extension. However, the Goodbye call handler greeting only announces the option to dial an extension. By default, the other three options are not announced during the greeting. https://ccieme.wordpress.com/category/ccnp-collaboration/capps/implementing-cisco-unity-connection-dial-plan-and-call-management/

    3. S

       spag22500

       2022-12-20

       2 upvotes

       B is correct: The Goodbye call handler allows the caller to sign in, restart the opening greeting, reach the operator, or dial an extension. However, the Goodbye call handler greeting only announces the option to dial an extension. By default, the other three options are not announced during the greeting. If the caller does not choose any option during that greeting (which is 5 seconds long), the call is terminated.

    4. T

       TexasGandalf

       2023-05-25

       Selected C2 upvotes

       Directory handler is the option that when created new, plays a default greeting that includes "from a touch tone phone you may dial an extension at any time."

18. Question 18 of 25

    When a user dials a number with a phone that is registered to the Cisco Unified Communications Manager, what is the default timeout before the number is sent?

    • A15 seconds
    • B5 seconds
    • C10 seconds
    • D3 seconds

    Show answer

    Correct answer: A

    How others answered

    • A71%
    • C29%

    Only 71% agreed on this answer — treat it as contested rather than settled.

    Explanation

    Please note that the document you are referring to is more than 14 years back (2006) CUCM v4, While we have and on hand proof that it is 15000 milliseconds. These questions supposed to reflect the latest CUCM versions 11.X or 12.X

    Adapted from community discussion by MrAshour · 2020-12-14 · 7 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. M

       MrAshour

       2020-12-14

       7 upvotes

       Please note that the document you are referring to is more than 14 years back (2006) CUCM v4, While we have and on hand proof that it is 15000 milliseconds. These questions supposed to reflect the latest CUCM versions 11.X or 12.X

    2. G

       Griswald

       2020-06-10

       5 upvotes

       NO SIP dial rules on phone: If the user dials digits but then does not press the Dial softkey or the # key, the phone will wait for inter-digit timeout (15 seconds by default) before sending a SIP INVITE message to Unified CM. For the example in Figure 14-8, dialing 1, 0, 0, 0 and waiting for inter-digit timeout would result in the phone placing a call to extension 1000 after 15 seconds.

    3. C

       CollabGuy

       2020-10-22

       4 upvotes

       It's 15s. If you don't believe, you can go to a fresh CUCM install (devnet sandbox for example): System => Service Parameters Select the CUCM server and the “Cisco CallManager” Service. Once there do search with “CTRL-F” to open up the find window in your browser. Then search for “302” This will bring you down to the “T302 Timer” settings field. You'll see that the default there will be 15000ms (15s)

    4. B

       b3532e4

       2024-09-29

       1 upvote

       When matching a variable-length pattern, the Cisco Unified Communications Manager does not know when dialing is complete, so it will wait for 15 seconds by default before processing the call. This post-dial delay can be reduced or eliminated as follows:  Reduce the Cisco CallManager service parameter called the T302 Timer to allow earlier detection of the end of dialing. However, do not set this timer to less than 4 seconds, to prevent premature transmission of the call before the user finishes dialing.  Configure a second route pattern, followed by the pound sign (#) wildcard, such as 9.011!# for North America or 0.00!# for Europe. Then educate users that they can indicate end of dialing by terminating the number with the # key. This action is analogous to pressing the Send button on a cell phone.

19. Question 19 of 25

    Select 2

    An engineer is integrating Unity Connection with Cisco UCM. Which two actions must be configured so that recording and playback from the IP phones works at all times, including peak traffic hours? (Choose two.)

    • AIncrease the number of voice ports.
    • BThe phone system to which the phones are registered in Unity Connection has the Default Trap Switch check box enabled.
    • CEnsure that you have set up SIP Digest Authentication on the SIP trunk security profile.
    • DIf it's a Unity Connection Cluster, ensure that replication is fine and not in split-brain mode.
    • EAdd dedicated dial-out ports with the allow trap connections setting selected.

    Show answer

    Correct answer: B, E

    How others answered

    • BE54%
    • AE31%
    • AD15%

    Only 54% agreed on this answer — treat it as contested rather than settled.

    Explanation

    Search for allow trap https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/integration/guide/cucm_sip/b_cucintcucmsip/b_cucintcucmsip_chapter_011.html#ID-2585-000003c7 Trap Connection needs to be enable on the port and phone system to always allow recording and playback

    Adapted from community discussion by kitty73 · 2024-02-05 · 4 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. K

       kitty73

       2024-02-05

       Selected B, E4 upvotes

       Search for allow trap https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/integration/guide/cucm_sip/b_cucintcucmsip/b_cucintcucmsip_chapter_011.html#ID-2585-000003c7 Trap Connection needs to be enable on the port and phone system to always allow recording and playback

    2. G

       G0y0

       2025-03-02

       Selected B, E1 upvote

       A. and D. are conditions that commit all of the ports (Answer Calls, Message Notification, MWI Requests, Allow TRAP Connections) and have an impact over all of the system, not just TRAP. Also ports are defined by the VM sizing. C. does not make any sense. B. and E. are more specific refering about the TRaP connections in order of the question. I think they are the most specific and precise about TRAP. The another three ones are generic and non-specific.

    3. C

       ciao

       2024-09-03

       1 upvote

       ++Voice Messaging Ports to Dial Out++ => Ports that only dial out and do not answer calls can do one or more of the following: - Notify users by phone, pager, or email of messages that have arrived. - Turn MWIs on and off for user extensions. - Make a TRAP connection so that users can use the phone as a recording and playback device in Unity Connection web applications.

    4. G

       G0y0

       2025-03-02

       1 upvote

       Reference: "Cisco Unified Communications Manager SIP Integration Guide for Cisco Unity Connection Release 12.x" https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/12x/integration/guide/cucm_sip/b_12xcucintcucmsip.pdf

20. Question 20 of 25

    

    Refer to the exhibit. An engineer configures ISDN on a voice gateway. The provider confirms that the PRI is configured with 10 channels the engineer ordered and is working from the provider side, but the engineer cannot get a B-channel to carry voice. The rest of the configuration for the serial interface and voice network is functioning correctly. Which actions must be taken to carry voice?

    • AThe engineer must activate the controller card on the voice gateway before configuring the device.
    • BThe engineer used a T1 interface but must use an E1 interface.
    • CThe pri-group timeslots command must be 0-9 for the 10 channels because all values on a router start with 0.
    • DThe engineer must manually revert the order of using the channels.

    Show answer

    Correct answer: B

    How others answered

    • B50%
    • D50%

    Only 50% agreed on this answer — treat it as contested rather than settled.

    Explanation

    D: is very likely the correct answer. A - is not correct, one cannot enter controller T1 interface if the Controller Card is not activated. B - is very unlikely. There is no indication of T1 vs E1, which normally only use outside of USA. C - is incorrect, the pri-group timeslot 1-10 cannot be 0-9 as "0" is not a valid number on Cisco IOS. D - since this is only fractional T1 circuit 1-10, by default IOS outbound calls are in descending order. All outbound calls will go out from bottom up(B-channel 23 and up) unless you manually configure the isdn busy b-channel XX to direct the IOS router not to sending calls out as a full T1 circuit. Therefore by manually revert the order of using the channels will work in this case. Understanding doing so may create glare but this is beside the point for this question.

    Adapted from community discussion by JC2008 · 2023-03-02 · 10 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. J

       JC2008

       2023-03-02

       10 upvotes

       D: is very likely the correct answer. A - is not correct, one cannot enter controller T1 interface if the Controller Card is not activated. B - is very unlikely. There is no indication of T1 vs E1, which normally only use outside of USA. C - is incorrect, the pri-group timeslot 1-10 cannot be 0-9 as "0" is not a valid number on Cisco IOS. D - since this is only fractional T1 circuit 1-10, by default IOS outbound calls are in descending order. All outbound calls will go out from bottom up(B-channel 23 and up) unless you manually configure the isdn busy b-channel XX to direct the IOS router not to sending calls out as a full T1 circuit. Therefore by manually revert the order of using the channels will work in this case. Understanding doing so may create glare but this is beside the point for this question.

    2. T

       Testme1235

       2023-02-25

       Selected B3 upvotes

       Option A, "The engineer must activate the controller card on the voice gateway before configuring the device," is not a valid solution to this issue because the controller card is already configured and functional since the voice network and serial interface are working correctly. The issue is related to the incorrect interface type being used, which is causing the B-channel to fail to carry voice. Therefore, activating the controller card will not resolve this particular issue.

    3. S

       sneff91

       2024-03-09

       3 upvotes

       Answer is indeed D. Cisco IOS default is a descending channel order for outbound calls: https://www.cisco.com/c/en/us/td/docs/ios/dial/configuration/guide/15-mt/dia_15_mt_book/dia_cfg_isdn_pri.html "Verify if the outgoing B channel calls are made in ascending or descending order. Cisco IOS default is descending order however, if the switch from the service providers is configured for outgoing calls made in ascending order, the router can be configured to match the switch configuration of the service provider."

    4. G

       G0y0

       2025-03-02

       1 upvote

       In fact, it does not matter the order of consumption, it does not define if a b channel exists or not. B,, believe ir or not, is the correct answer. VWIC1/2/3 all of them support E1/T1. If the carrier has a E1configured and the customer has a T1 configured, is enough reason to do not have b-channels, but yo can have d-channel because d-channel is into Layer 2. Commonly, you must ensure that the framing, line coding, and clock source are configured correctly. And the way that you could have this issue, this is one, that customer configured card type t1 meanwhile in the other side have e1, then you will have problem with framing and line code. So, change card type from t1 to e1, and configure a E1 with the correct framing and linecode. B, is correct. https://www.cisco.com/c/en/us/support/docs/wan/t1-e1-t3-e3/8131-T1-pri.html https://www.cisco.com/c/en/us/support/docs/wan/t1-e1-t3-e3/10205-t1-flchrt-main.html

21. Question 21 of 25

    How does WRED prevent congestion?

    • AIt marks which packets to drop randomly.
    • BIt drops packets selectively based on the IP precedence value.
    • CIt marks which packets to drop based on how much traffic the client sent.
    • DIt drops packets that have a higher DSCP value last.

    Show answer

    Correct answer: B

    How others answered

    • B100%

    Explanation

    Answer B About Weighted Random Early Detection Random Early Detection (RED) is a congestion avoidance mechanism that takes advantage of the congestion control mechanism of TCP. By randomly dropping packets prior to periods of high congestion, RED tells the packet source to decrease its transmission rate. WRED drops packets selectively based on IP precedence. Edge routers assign IP precedences to packets as they enter the network. (WRED is useful on any output interface where you expect to have congestion. However, WRED is usually used in the core routers of a network, rather than at the edge.) WRED uses these precedences to determine how it treats different types of traffic. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/qos_conavd/configuration/xe-16/qos-conavd-xe-16-book/qos-conavd-cfg-wred.html

    Adapted from community discussion by gorzad · 2024-06-30 · 2 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. G

       gorzad

       2024-06-30

       2 upvotes

       Answer B About Weighted Random Early Detection Random Early Detection (RED) is a congestion avoidance mechanism that takes advantage of the congestion control mechanism of TCP. By randomly dropping packets prior to periods of high congestion, RED tells the packet source to decrease its transmission rate. WRED drops packets selectively based on IP precedence. Edge routers assign IP precedences to packets as they enter the network. (WRED is useful on any output interface where you expect to have congestion. However, WRED is usually used in the core routers of a network, rather than at the edge.) WRED uses these precedences to determine how it treats different types of traffic. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/qos_conavd/configuration/xe-16/qos-conavd-xe-16-book/qos-conavd-cfg-wred.html

    2. G

       G0y0

       2025-03-03

       Selected B1 upvote

       Weighted random early detection (WRED) is a Cisco-proprietary congestion avoidance technique. WRED increases the probability that congestion (that is, an output interface queue completely filling to capacity) is avoided by dropping low-priority packets rather than dropping high-priority packets. selectively drop packets when buffers are filling up. This random drop is used to avoid congestion. "Implementing Cisco Unified Communications Voice over IP and QoS (CVOICE) Foundation Learning Guide"

    3. D

       DaKenjee

       2024-05-28

       Selected B1 upvote

       Answer B Answer A could not fit, it poorly explains on which decision/based on it marks traffic WRED uses these precedences to determine how it treats different types of traffic. https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/qos_conavd/configuration/15-mt/qos-conavd-15-mt-book/qos-conavd-cfg-wred.html

    4. M

       Mert_kerna

       2024-06-01

       Selected B1 upvote

       In fact, RED (random early detection) randomly drops packets. WRED (weighted random early detection) literally selectively discards lower priority traffic. With WRED, IP Precedence governs which packets are dropped. So the answer 100% has to be "B".

22. Question 22 of 25

    

    Refer to the exhibit. A Cisco UCM user with directory number 4401 dials 5507, and the call is routed to a Cisco Unified Border Element. Which IP address will the call be sent to?

    • A10.0.0.2
    • B10.0.0.3
    • C10.0.0.4
    • D10.0.0.1

    Show answer

    Correct answer: A

    How others answered

    • A75%

    Explanation

    Actually the correct answer is A, voice class dpg Completely ignores destination-pattern configured in the dial-peer section and follows preference. To be sure I build this setup in my lab which confirms my statement.

    Adapted from community discussion by Nicetomeetyou · 2021-07-01 · 18 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. N

       Nicetomeetyou

       2021-07-01

       18 upvotes

       Actually the correct answer is A, voice class dpg Completely ignores destination-pattern configured in the dial-peer section and follows preference. To be sure I build this setup in my lab which confirms my statement.

    2. M

       MKZ

       2021-06-16

       7 upvotes

       incoming call first hit dial-peer 1001 -> voice class DGP 2000 > Preference 2 I think 10.0.0.3.

    3. D

       davidanibalmarcelino

       2021-06-16

       5 upvotes

       if it was not for the voice class dpg 2000 it would be 10.0.0.4. I tested on my lab gateway and you are right, 10.0.0.3 is preferred.

    4. K

       Komy

       2021-06-30

       4 upvotes

       10.0.0.3 is correct Snippet from Cisco: "Once an incoming call is matched by an inbound dial peer with an active destination dial-peer group, dial peers from this group are used to route the incoming call. No other outbound dial-peer provisioning to select outbound dial peers is used. A preference can be defined for each dial peer in a dial-peer group"

23. Question 23 of 25

    Refer to the exhibit.

    

    An administrator is attempting to resolve a DTMF issue to a specific IVR and has supplied logs from the most recent issue. What is the cause and solution of the issue?

    • AThe local endpoint supports RFC2833, and the IVR supports only KPML. Configure an MTP to resolve the DTMF mismatch.
    • BThe local endpoint does not support RFC2833, and the IVR supports only RFC2833. Configure a transcoder to resolve the DTMF mismatch.
    • CThe local endpoint does not support RFC2833, and the IVR supports only RFC 2833. Configure an MTP to resolve the DTMF mismatch.
    • DThe local endpoint supports RFC2833, and the IVR supports only KPML. Configure a transcoder to resolve the DTMF mismatch.

    Show answer

    Correct answer: C

    How others answered

    • C57%
    • A43%

    Only 57% agreed on this answer — treat it as contested rather than settled.

    Explanation

    I would say it's C. mLocalDtmfCaps has KPML=1, Inband=0, so that means the local endpoint does not support RFC2833. mEndpointsDtmfCaps would be the IVR and it has the opposite (KPML=0, Inband=1). For that scenario I think an MTP would be the way to go, not an transcoder: "C. Need MTP In this scenario, SCCP EP supports OOB only, and SIP EP supports RFC2833 only. Therefore an MTP is needed. MTP will send\receive RFC2833 packets to\from the SIP EP and will send\receive OOB DTMF packets to\from CCM. CCM will send\receive OOB DTMF packets to\from MTP and the SCCP phone." https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/118708-technote-dtmf-00.html

    Adapted from community discussion by TheBabu · 2024-04-25 · 3 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. T

       TheBabu

       2024-04-25

       Selected C3 upvotes

       I would say it's C. mLocalDtmfCaps has KPML=1, Inband=0, so that means the local endpoint does not support RFC2833. mEndpointsDtmfCaps would be the IVR and it has the opposite (KPML=0, Inband=1). For that scenario I think an MTP would be the way to go, not an transcoder: "C. Need MTP In this scenario, SCCP EP supports OOB only, and SIP EP supports RFC2833 only. Therefore an MTP is needed. MTP will send\receive RFC2833 packets to\from the SIP EP and will send\receive OOB DTMF packets to\from CCM. CCM will send\receive OOB DTMF packets to\from MTP and the SCCP phone." https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/118708-technote-dtmf-00.html

    2. G

       G0y0

       2025-03-04

       1 upvote

       Now, it is necessary to question, what is needed to DTMF interworking between in-band and OOB? To interworking DTMF, it is needed to use MTP in a supported codec of course, but in general, to do DTMF interworking essencially it is the reason to be of the MTP. If there is any codec mismatch, transcoding enters in scene, but it is other stuff. So, in order of the above, correct answer is C. I though the solution was B, however, reviewing the following reference, I realized that C is the correct answer. "Understanding DTMF negotiation and troubleshooting on SIP Trunks" https://community.cisco.com/t5/collaboration-knowledge-base/understanding-dtmf-negotiation-and-troubleshooting-on-sip-trunks/ta-p/3152511 "Configure DTMF Relay on CUBE" https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#toc-hId-996216602

    3. G

       G0y0

       2025-03-04

       Selected C1 upvote

       *By endpoints here I mean a Phone/IVR/voicemail application and the gateway/trunk. *The "mLocalDtmfCaps" refers to the dtmf methods supported by CUCM. Then: A. and D are incorrect.The local endpoint (CUCM in behalf of the phone) does not support RFC2833 here.

    4. R

       rick2

       2025-02-01

       1 upvote

       C is correct. https://community.cisco.com/t5/collaboration-knowledge-base/understanding-dtmf-negotiation-and-troubleshooting-on-sip-trunks/ta-p/3152511

24. Question 24 of 25

    A collaboration engineer adds a voice gateway to Cisco UCM. The engineer creates a new gateway device in Cisco UCM, selects VG320 as the device type, and selects MGCP as the protocol. What must be done next to add the gateway to the Cisco UCM database?

    • ASelect a device pool for the new gateway.
    • BAdd the FQDN or hostname of the device.
    • CConfigure the module in slot 0 of the new gateway.
    • DSelect the DTMF relay type for the gateway.

    Show answer

    Correct answer: B

    How others answered

    • B67%
    • C33%

    Only 67% agreed on this answer — treat it as contested rather than settled.

    Explanation

    Answer is B: This snippet is captured from the link below: "Now that you have selected the hardware and protocol used, you need to configure the Domain Name, Cisco Unified Communications Manager Group, and the Module Information. These are the major fields that are required to register an endpoint via MGCP." So Cisco has listed the order as [Select Hardware, Protocol then FQDN/Hostname]. in the question, we can see it already told you that Hardware is: VG320, Protocol is :MGCP , the next step has to be Specify FQDN/Hostname https://www.cisco.com/c/en/us/support/docs/voice/media-gateway-control-protocol-mgcp/214635-configure-and-troubleshoot-mgcp-gateways.html

    Adapted from community discussion by Komy · 2024-10-17 · 5 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

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    1. K

       Komy

       2024-10-17

       Selected B5 upvotes

       Answer is B: This snippet is captured from the link below: "Now that you have selected the hardware and protocol used, you need to configure the Domain Name, Cisco Unified Communications Manager Group, and the Module Information. These are the major fields that are required to register an endpoint via MGCP." So Cisco has listed the order as [Select Hardware, Protocol then FQDN/Hostname]. in the question, we can see it already told you that Hardware is: VG320, Protocol is :MGCP , the next step has to be Specify FQDN/Hostname https://www.cisco.com/c/en/us/support/docs/voice/media-gateway-control-protocol-mgcp/214635-configure-and-troubleshoot-mgcp-gateways.html

    2. S

       skneff

       2024-07-12

       Selected B1 upvote

       The answer is B. https://community.cisco.com/t5/collaboration-knowledge-base/mgcp-gateway-integration-with-cucm-and-pstn-service-provider/ta-p/3116588#toc-hId-1719765046 "Step1: Enter the host name or fully qualified domain name (FQDN) of the gateway in the Domain name field. If a domain name is specified in the Cisco router you must provision an FQDN. If the FQDN specified in the CUCM does not match the “host name” dot “domain name” of the Cisco IOS router, then end point will never register. A router with the host name of Router1 and domain name of highpoint.com would have an FQDN of Router1.highpoint.come" You can't use the IP address of the gateway either, so as written, B is correct. You need to configure a module to get the ports themselves to register, but not for the gateway to be simply entered into the database, which is what the question is asking.

    3. T

       TheBabu

       2024-10-23

       Selected B2 upvotes

       Should be B, when you add a new gateway, and fill in the fields in the configuration page in order, the domain name is the first required item to fill.

    4. M

       MaxG

       2023-12-01

       Selected B2 upvotes

       B is correct per https://community.cisco.com/t5/collaboration-knowledge-base/mgcp-gateway-integration-with-cucm-and-pstn-service-provider/ta-p/3116588

25. Question 25 of 25

    Select 2

    Endpoint A is attempting to call endpoint B. Endpoint A only supports G.711ulaw with a packetization rate of 20 ms, and endpoint B supports packetization rate of 30 ms for G.711ulaw. Which two media resources are allocated to normalize packetization rates through transrating? (Choose two.)

    • Asoftware transcoder on Cisco UCM
    • Bhardware transcoder on Cisco IOS Software
    • Csoftware MTP on Cisco IOS Software
    • Dsoftware MTP on Cisco UCM
    • Ehardware MTP on Cisco IOS Software

    Show answer

    Correct answer: D, E

    How others answered

    • DE52%
    • BE29%
    • BD19%

    Only 52% agreed on this answer — treat it as contested rather than settled.

    Explanation

    It's B and E. This can only be done by using DSPs. Hardare MTP and Transcoder through a router. CUCM doesn't have a software transcoders so for sure A cannot be the answer. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/media.html#77886

    Adapted from community discussion by gahhhhaccount · 2021-06-11 · 14 community upvotes. Third-party contribution; reviewed by our quality filter, not independently verified.

    Supporting references

    • Cisco documentation
    • Cisco documentation
    • Cisco documentation
    • Cisco documentation

    View community discussion4

    Candidate context

    Question 25 discussion

    These are quality-filtered excerpts from third-party community discussion. They can be useful reasoning signals, but they are not official answers and have not been independently verified by 350-801 or CertExamHQ.

    1. G

       gahhhhaccount

       2021-06-11

       14 upvotes

       It's B and E. This can only be done by using DSPs. Hardare MTP and Transcoder through a router. CUCM doesn't have a software transcoders so for sure A cannot be the answer. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/media.html#77886

    2. A

       asdlfhqwoiefnwe

       2022-01-06

       Selected D, E9 upvotes

       D and E While a transcoder could work, MTP is sufficient and the better answer. CUCM and IOS have MTP resources, but IOS can only handle transrating when using DSPs (which would be configured as a hardware MTP).

    3. S

       SDLOA14

       2021-06-19

       6 upvotes

       B and E Agree From Cisco SRND 11 "When Cisco IOS MTP resources are invoked by Unified CM for a call flow, a software session rather than a hardware DSP session is consumed unless the media legs of the call flow require transrating. Thus, for flows invoking an MTP, a DSP session [hardware MTP] is used only when transrating (conversion between media legs with the same codec but different packetization times) is required."

    4. [

       [Removed]

       2022-06-28

       Selected D, E2 upvotes

       DE. Repacketization needs a DSP and cannot be done by software only. this means its obviously E on iOS, but also D, because on CUCM a hardware MTP is also refered to as a software MTP (https://www.cisco.com/c/en/us/td/docs/ios/voice/cminterop/configuration/guide/15_1/vc_15_1_book/vc_enh_confr_vgr.html) The following MTP resources are supported for Cisco Unified Communications Manager 4.0 (formerly known as Cisco CallManager 4.0) and later releases: •Software MTP—Software-only implementation that does not use a DSP resource for endpoints using the same codec and the same packetization time. •Hardware MTP—Hardware-only implementation that uses a DSP resource for endpoints using the same G.711 codec but a different packetization time. >>>> >>The repacketization requires a DSP resource so it cannot be done by software only. Cisco Unified Communications Manager also uses the term software MTP when referring to a hardware MTP. <<<<<<

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