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Here you will find answers to CVoice – Call Routing and Path Selection Questions
Question 1
When setting up a VoIP call, what is the first thing a gateway router tries to match to a dialed number?
A. call leg
B. IP route
C. session target
D. destination pattern
Answer: D
Explanation
First, the gateway attempts to match the called number with the incoming called-number. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the answer-address of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the destination-pattern of each dial-peer.
Notice that these steps are just applied for inbound dial peer.
Question 2
Refer to the exhibit. Highland Park Property Development is integrating a Cisco Unified Communications Manager Express system with the existing PBX via an E1 QSIG trunk. After the initial configuration, no calls can be placed from IP phones to PBX phones. How can this problem be resolved?
1d20h: ISDN Se3/0:15: Outgoing call id = 0x85F4, dsl 0
1d20h: ISDN Se3/0:15: process_pri_call(): call id 0x85F4, number 35293315, speed 0, call type VOICE, redialed? f, csm call? f, pdata? t
1d20h: callED type/plan overridden by call_decode
ld20h: did’t copy oct3a reason: not CALLER_NUMBER_IE
ld20h: building outgoing channel id for call nfas_int is 0 len is 0
ld20h: ISDN se3/0:15: TX -> INFOc sapi = 0 tei =0 ns = 19 nr = 19 i =
0x080200890504038090A31803A983811E0281837009803335323933333135
ld20h: SETUP pd = 8 callref = 0x0089
ld20h:    Bearer capability i = 0x8090A3
ld20h:    Channel id i = 0XA98381
ld20h:    Progress Ind i =0x8183 – Origination address is non-ISDN
ld20h:    called Party Number i = 0x80, ‘35293315’, Plan:unknown, Type:unknown
ld20h: ISDN Se3/0:15: RX <- RRr = 0 tei = 0 nr = 20
Id20h: ISDN se3/0:15: RX <- INFOc sapi = 0 tei =0 ns = 19 nr = 20 i = 0x080280895A08028286
ld20h: RELEASE_COMP pd = 8 callref = 0x8089
ld20h:    Cause i = 0x8286 – Channel unacceptable
ld20h: ISDN Se3/0:15: TX -> RRr sapi = 0 tei =0 nr = 20
ld20h: ISDN se3/0:15: CCPRI_Releasecall(): bchan 1, call id 0x85F4, call type VOICE
ld20h: ccPRI_ReleaseChan released b_dsl 0 B_chan 1
ld20h: ISDN Se3/0:15: LIF_EVENT: ces/callid l/0x85F4 CALI__REJECTION
ld20h: ISDN Se3/0:15: LIF_EVENT: ces/callid 1/0x85F4 CALL_CLEARED
ld20h: ISDN Se3/0:15: received CALL_CLEARED call_id 0x85F4
A. Increase the ISDN T302 timer to allow more time for call setup.
B. Add the command isdn negotiate-bchan to the serial interface.
C. Add the command isdn contiguous-bchan to the serial interface.
D. Change the channel selection order from descending to ascending.
Answer: B
Question 3
Refer to the exhibit. The Carmichael caller dials the site access code for Merrimack (6) followed by the four digit extension number of the destination phone (0124). If the call is going to go across the IP WAN, which action will have to be taken?
IPWAN_Routing.jpg
A. Translate 60124 to 5125550124.
B. Strip the site access code and send four digits.
C. Strip the site access code and prepend 1512555.
D. Do nothing because the site access code matches the last five digits of the target number.
E. Strip the site access code, send four digits, then prepend the access code when it reaches the Merrimack gateway.
Answer: B
Explanation
The site access code (6) is just used to inform the originating gateway which gateway it needs to send traffic to. Therefore, after learning the traffic should be sent to Merrimack gateway, it trips off the site access code. Notice that the receiving gateway will receive “0124”, which is enough information to ring the phone plugged into it.
Question 4
Which path selection mechanism lets you choose either the even or odd channels first?
A. hunt groups
B. trunk groups
C. tailend hopoff
D. Call Admission Control
Answer: B
Explanation
By using trunk groups, we can choose to use either the even or odd channels first with the command:
hunt-scheme …. [even | odd ….] (notice: the full command is very long so I shorten it to the simplest form)
Question 5
When using CUBE, which two statements describe how media flow-through differs from media flow-around? (Choose two.)
A. Media flow-around provides address hiding by terminating both signaling and RTP streams.
B. Media flow-through terminates the signaling channel and the RTP streams flow directly between endpoints.
C. Media flow-around and media flow-through function in a similar manner, but media flow-around supports NAT traversal.
D. Media flow-through terminates the RTP streams but allows signaling to flow directly between endpoints.
E. Media flow-around terminates the signaling stream and allows RTP streams to flow directly between endpoints.
F. Media flow-through provides address hiding by terminating both signaling and RTP streams.
Answer: E F
Explanation
Media flow through and media flow around mode is supported on the Cisco Unified Border Element (CUBE). The CUBE is always involved in the call setup (signaling) portion of the call, but the media (RTP bearer stream) may flow through the CUBE or be routed around the platform.
Media flow around allows the CUBE greater scalability in the number of calls that can be processed by one CUBE router.
For Media flow through option, the media packets are passed through the CUBE, they will get terminated and re-originates with CUBE’s IP address and port number, so here we cannot find the original gateway’s ip address. This is one of the security feature in the CUBE. The default option is “media flow-through”.
Question 6
Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?
dial-peer voice 1 pots
translation-profile incoming in1


trunk group 101
carrier-id 1642
hunt-scheme sequential even up
translation-profile incoming in1

controller T1 1/0
ds0-group 1 timeslots 1-24 type e&m-fgd
cas-custom 1
trunk-group 101

voice-port 1/0
translation-profile incoming in1
trunk-group 101 1

voice service pots
translation-profile incoming controller T1 1/0 in1
A. The call will be routed to the longest idle channel.
B. The call will be routed to the least used channel.
C. The call will be routed to a random available channel.
D. The call will be routed to the next available channel, starting from channel 1, hunting up toward channel
E. The call will be routed to the next available channel, starting from channel 24, hunting down toward channel 1.
Answer: E
Explanation
In the configuration, we learn that the hunt-scheme sequential is used. It specifies the sequential search method for finding an available channel in a trunk group for outgoing calls. The syntax of this command is shown below:
hunt-scheme sequential [both | even | odd [up | down] ]
Description:
both: Searches both even and odd numbered channels.
even: Searches for an idle even numbered channel. If no idle even numbered channel is available, an odd-numbered channel is sought.
odd: Searches for an idle odd numbered channel. If no idle odd numbered channel is available, an even-numbered channel is sought.
up: Searches channels in ascending order based within a trunk group member.
down: Searches channels in descending order within a trunk group member.
Notice that up down parameters are used with botheven or odd.
Therefore the command hunt-scheme sequential even up searches in ascending order for an even numbered idle channel starting with the trunk group member of highest precedence. I am not so sure but channel 24 will have highest precedence so the “hunt” begins from channel 24 down to channel 1. Therefore, E is the most suitable solution for this question.
The cas-custom command is used to customize T1/CAS signaling parameters for a particular T1 channel group on a channelized T1 line.
Question 7
Site A uses three-digit internal numbers and remote Site B uses four-digit internal numbers. All calls to the PSTN are routed through Site B. What dial plan below best represents provision simplicity, assuming the NANP numbering plan?
A. Translate all called numbers within Site A to four digits.
B. Translate all called numbers within Site B to three digits.
C. Translate all called numbers leaving Site A to ten digits.
D. Translate all called numbers at either site to ten digits.
Answer: C
Explanation
North American Numbering Plan (NANP) is designed around a 10-digit numbering plan:
NANP_tendigits.jpg
(Sometimes you will see it as NXX – NXXX – XXXX, which means that the first and fourth digits can’t be zero or one)
It consists of 3-digit area codes and 7-digit telephone. For telephone numbers that are located within an area code, the PSTN uses a seven-digit dial plan numbers.
Notice that “Site B uses four-digit internal numbers” means we need ten digits to access site B from an outside PSTN. Therefore, if people from Site A want to call people at site B and sometimes they just press 4 digits then the administrators should translate the called numbers to ten digits before leaving Site A.

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