Cisco Voice over IP

Size: px
Start display at page:

Download "642-436 - Cisco Voice over IP"

Transcription

1 Cisco Voice over IP Number: Passing Score: 825 Time Limit: 120 min File Version: This material is copy from pass4sure All answers was collected by print screens. Some answer are correcteds by research on

2 Exam A QUESTION 1 Which two statements describe the purpose of the technology prefix? (Choose two). A. Technology prefixes are configured on gateways to indicate to the gatekeeper whether they support voice or video. B. Technology prefixes must always be configured on gateways. C. Technology prefixes are uses to indentify different types or classes of gateways. D. Technology prefixes are prepended to the destination address by the gateway. E. Technology prefixes have to be unique on each gateway. Correct Answer: CD /Reference: QUESTION 2 Which statement is true about only out-of-band signaling? A. A signaling bit is robbed from each frame B. Signaling bits are sent in a special order in a dedicated signaling frame. C. All signaling is directly associated with its corresponding voice frame. D. All voice pacckets carry their own signaling. Correct Answer: B /Reference: QUESTION 3 Examine the example output. hostname GW1! interface Ethernet 0/0 ip address h323-gateway voip interface h323-gateway voip id GK1-zone1.abc.com ipaddr h323-gateway voip h323-id GW1 h323-gateway voip bind srcaddr ! dial-peer voice 1 voip destination-pattern session-target ras! dial-peer voice 2 pots destination-pattern no register e164! end

3 Choose the command that will restore communication with gatekeeper functionality to this device. A. h323-gateway voip h323-id GK1 B. gateway C. h323-gateway voip bind srcaddr D. h323-gateway voip GW1-zone2.abc.com abc.com ipaddr Correct Answer: B /Reference: QUESTION 4 Refer to the exhibit. Which actions would be initiated by a UAS? (Choose two.) A. contacts the user when a SIP invitation is received. B. originates the REFER method to initiate call termination. C. returns a response on behalf of the user to the invitation originator. D. originates the INVITE method including a description of the session parameters. E. originate the ACK method to indicate that it has received a response to an invitation. Correct Answer: AC /Reference: QUESTION 5 Which option is true concerning the MGCP call agent? A. acts only as recorder of call details B. provides only call signaling and call setup C. manages all aspects of the call and voice stream D. monitors the quality of each call after setup Correct Answer: B

4 /Reference: QUESTION 6 Which statement best describes gatekeeper operation when the technology prefix is matched and the gatekeeper is using the technology prefix with hopoff? A. The gatekeeper always forwards the call to the zone specified in the hopoff command. B. The gatekeeper only forwards the call to the hopoff zone if the zone prefix does not match. C. The gatekeeper attemps to forward the call to the hopoff zone, but if this fails, it will forward the call to the zone specified in the zone prefix command. D. The gatekeeper attempts to forward the call to the zone specified in the zone prefix command first, but if this fails, it will forward the call to the zone specified in the hopoff command. Correct Answer: A /Reference: QUESTION 7 Refer to the exhibit. All IP phones are sccp phones. Phone D makes an internal call to phone G. Which call setup signaling statement is true? A. Phone D signals phone G directly. Call setup is handled by the phones. B. Phone D signals gateway A, which processes the call and signals phone G.

5 C. Phone D signals gateway B, which processes the call and signals phone G. D. Phone D signals gatekeeper. The gatekeeper processes the call and signals phone G. E. Phone D signals the call agent. The call agent processes the call and signals phone G. Correct Answer: E /Reference: QUESTION 8 Refer to the exhibit. All IP phones use SCCP. Fax machine F calls fax machine J. Which call setup signaling statement is correct? A. Fax F signals Fax J directly. Call setup is handled by the fax machines. B. Gateway A processes the call and signals gateway B. Gateway B processes the request. C. Gateway A signals the call agent. The call agent processes the call and signals gateway B. D. Gateway A signals the gatekeeper. The gatekeeper processes the call and signals gateway B. E. Gateway A processes the call and signals gateway B. Gateway B processes the call. During the setup, the gateways query the gatekeeper for address resolution and call setup permission. Correct Answer: E /Reference: QUESTION 9 Refer to the exhibit. Three department managers share the directory number The Marketing manager s phone is attached to port 1/1. The Engineering manager s phone is attached to port 1/2. The Shipping manager s phone is attached to port 1/3. In which situation would an incoming call ring on the Shipping manager s phone?

6 A. The Marketing manager is on the phone. B. None of the managers are on the phone. C. The Engineering manager is on the phone. D. The Shipping manager and Marketing manager are on the phone E. The Engineering manager and Marketing manager are on the phone. Correct Answer: E /Reference: QUESTION 10 Using Cisco Unified Communications Manager Express, what four steps are necessary to implement COR? (Choose four.) A. Configure SRST. B. Define COR labels. C. Configure COR lists. D. Assign COR list to ephone-dn. E. Configure COR lists on the voice ports. F. Configure diall peers and assign COR lists. Correct Answer: BCDF /Reference: QUESTION 11

7 At what point does the MGCP call agent release the setup of the call path to the residential gateways? A. After the call agent has been notified that an event occurred at the source residential gateway. B. After the call agent has been notified of an event and has instructed the source residential gateway to create a connection. C. Does not release call path setup. D. After the call agent has sent a connection request to both the source and destination and has relayed a modify-connection request to the source so that the source and destination can set up the call path. E. After the call agent has forwarded session description protocol information to the destination from the source and has sent a modify connection to the destination and a create-connection request to the source. Correct Answer: D /Reference: QUESTION 12 Which two codes together make up the number that follows the E.164 recommendation numbering scheme? (Choose two.) A. Country code B. subscriber code C. national destination code D. provider code Correct Answer: AB /Reference: QUESTION 13 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. Correct Answer: EF /Reference: QUESTION 14

8 Refer to exhibit. The Acme Corp. uses H.323 to place calls to their supplier RR industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE? A. service voice voip allow-connections h323 to h323 allow-connections h323 to sip B. voice service voip allow-connections h323 to h323 allow-connections h323 to sip C. voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 D. service voice voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to sip allow-connections sip to h323 Correct Answer: C /Reference: QUESTION 15 Which two are types of Call Admission Control? (Choose Two.) A. local B. QoS-based C. resource-based D. topology-based E. gateway zone bandwidth F. gatekeeper-controlled RSVP Correct Answer: AC

9 /Reference: QUESTION 16 Using a standalone IOS gateway, which three steps are necessary to implement COR (Choose three.) A. Configure SRST. B. Define COR labels. C. Configure COR lists. D. Assign COR list to ephone-dn. E. Configure COR lists on voice ports. F. Configure dial peers and assign COR lists. Correct Answer: BCF /Reference: QUESTION 17 Refer to the exhibit. You have configured a gatekeeper and an IP-IP gateway on the same router. When you look at the output from the show gatekeeper endpoint command, the IP-IP gateway is not registered with the gatekeeper. What needs to be configured to resolve this issue?

10 A. You need stop and restart the gateway. B. You need to add a Voip dial peer to the configuration. C. The H323-gateway voip id command has a incorrect IP address. D. The H323-gateway voip id command has a incorrect gatekeeper ID and IP address. Correct Answer: B /Reference: QUESTION 18 Which process changes an internal extension into a fully qualified external PSTN number before matching to a dial peer? A. digit masking

11 B. forward digits C. number expansion D. prefix extension Correct Answer: C /Reference: QUESTION 19 The SJ local zone contains a gatekeeper that controls two gateways, SJ1 and SJ2. Both gateways provide access to area code 408. Which two command strings should be entered into the gatekeeper to give the SJ2 gateway priority over the SJ1 gateway? (Choose two.) A. zone prefix SJ 408 gw-priority 6 SJ1 B. zone prefix SJ 408 gw-priority 6 SJ2 C. zone prefix SJ 408 gw-priority 10 SJ1 D. zone prefix SJ 408 gw-priority 10 SJ2 E. zone prefix SJ 408 gw-priority 0 SJ2, 10 SJ1 F. zone prefix SJ 408 gw-priority 6 SJ1, 10 SJ2 Correct Answer: AD /Reference: QUESTION 20 Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk? A. voice-port 1/0:1 B. connection trunk C. ds0-group timeslots 1-23 type ext-sig D. connection trunk answer-mode E. connection-trunk answer-mode Correct Answer: E /Reference: QUESTION 21 Refer to the exhibit. Choose the correct configuration command set that will allow the gateway in zone BR to register with the gatekeeper at HQ so that the gateway is placed in zone BR.

12 A. B. C. D.

13 Correct Answer: C /Reference: QUESTION 22 Which best defines an ACD? A. a telephone system that switches calls between users on local lines B. a local company that provides phone capability and distribution from the phone company s central office. C. a telephone system that responds to a caller with a voice menu and helps to appropriately connect the call D. a telephone system that is connected to the exchange to provide conventional voice services to several subscribers Correct Answer: C /Reference: QUESTION 23 You have been asked to deploy a gatekeeper to support CUBE that will connect your organizational domain to the domain of an Internet Telephony Service Provider so that callers can reach the 407 area code. Which configuration will support this function? A. B. C.

14 D. Correct Answer: D /Reference: QUESTION 24 What is the E.164 standard? A. dial plan B. private numbering plan C. national numbering plan D. international public telecommunications numbering plan Correct Answer: D /Reference: QUESTION 25 You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule command? A. test translation-rule The replaced number: B. test translation-rule The replaced number: C. test translation-rule The replaced number: D. test translation-rule 1 910

15 The replaced number: Correct Answer: A /Reference: QUESTION 26 A customer needs to configure a CAS E & M circuit that will support inbound and outbound DNIS and inbound ANI. Which configuration will accomplish this task? A. pri-group timeslots 1-24 B. ds0-group 0 timeslots 1-24 type none C. ds0-group 0 timeslots 1-24 type e&m-fgd D. ds0-group 0 timeslots 1-24 type fgd-eana E. ds0-group 0 timeslots 1-31 type r2-digital r2-digital-compelled ani Correct Answer: C /Reference: QUESTION 27 To hide identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B? A. proxy B. redirect C. registrar D. user agent client E. user agent server Correct Answer: A

16 /Reference: QUESTION 28 Which dial-peer command can set the parameters that search through a series of dial peers for a destination that is not in use? A. hunt B. query C. rotary D. request E. circulate F. distribute Correct Answer: A /Reference: QUESTION 29 Refer to the H.323 message in the exhibit. What is the gateway doing with the gatekeeper?

17 A. initial registration B. full registration C. lightweight registration D. registration retry Correct Answer: C /Reference: QUESTION 30 A customer wants to roll out IP telephony to the regional office. They are currently using the G.711 codec at headquarters Which codec will support voice activity detection and comfort noise generation? A. G.711 B. G.726 C. G.729B D. G Correct Answer: C

18 /Reference: QUESTION 31 Which three services are supported by CUBE when supporting H323-to-SIP calls? (Choose three.) A. SIP causes code B. media flow-around C. media flow-through D. codec transparent support E. Transport Layer Security F. H.261, H.263 and H.264 video codecs Correct Answer: CDE /Reference: QUESTION 32 A B C D Correct Answer: A /Reference: QUESTION 33

19 Select and Place: Correct Answer:

20 /Reference:

21 QUESTION 34 Refer to exhibit. What is the minimum WAN bandwith required to support three simultaneous VOIP calls in this network?

22 A. 19,200 bps B. 51,600 bps C. 79,200 bps D. 247,200 bps Correct Answer: C /Reference: QUESTION 35 Where would you assign COR lists in Cisco Unified Communications Manager Express? A. ephone B. ephone-dn C. voice register dn D. voice register pool Correct Answer: B /Reference: QUESTION 36 What does a gatekeeper do when it matches a technology prefix? A. strips off the technology prefix and sends the matching zone prefix to the remote gatekeeper B. send both the technology prefix and zone prefix to the remote gatekeeper C. strips off the zone prefix and forwards the tecnology prefix to the remote gatekeeper D. strips off both the technology prefix and zone prefix and forwards the remaining destination number Correct Answer: B /Reference:

23 QUESTION 37 Which CUBE configuration will support H.323 protocol interworking and address hiding? A. voice services voip h323 interworking media flow-around B. voice services h323 to h323 h323 interworking media flow-through C. voice services voip all-connections h323 to h323 media flow-around D. voice service voip allow-connections h323 to h323 Correct Answer: D /Reference: QUESTION 38 Which statement is true about MGCP? A. Call completion is always shared, with some intelligence on the endpoint, some on the call agent B. Endpoints always take all actions to complete calls. C. Endpoints may act alone or cooperate with call agent to complete calls. D. Call agents order and direct each step of call completion for the endpoints. Correct Answer: D /Reference: QUESTION 39 A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined two translation profiles, one for incoming and one for outgoing calls. What can be used to simplify this task? A. dial peer B. voice port C. hunt group D. trunk group E. source IP group Correct Answer: D /Reference: QUESTION 40

24 Refer to the exhibit. Your customer has connected an existing PBX to the IP network. The PBX users can make calls to other extensions on the PBX but are unable to call the test extension All others applications on the IP network are working correctly. Compare the PBX system requirements to the configuration for R1 in the exhibit. Which configuration change will resolve the problem? A. configure forward digits all in dial-peer 1 POTS B. configure wink-start signaling on voice-port 1/1/0 C. configure operation 4-wire and type 5 on voice-port 1/1/0 D. configure operation 2-wire and type 5 on voice-port 1/1/0 Correct Answer: C /Reference: QUESTION 41 Refer to the output from the debug h225 asn1 command in the exhibit. You have configured a gatekeeper with two local zones, hq and br. You want the gateway at the branch location to register with zone BR. What needs to be corrected in the branch gateway to resolve the issue?

25

26 A. Change the IP address in the h323-gateway voip id command. B. Change the gatekeeper-id in the h323-gateway voip id command C. Add a zone remote for zone BR so the gateway can register with the correct zone D. Change the gatekeeper-id and the IP address in the h323-gateway voip id command. Correct Answer: B /Reference: QUESTION 42 A. LFI B. RTP C. UDP D. RTCP E. CRTP Correct Answer: B /Reference: QUESTION 43

27 Select and Place: Correct Answer:

28 /Reference: QUESTION 44

29 A. The outbound VoiP diall peer is matched and all digits are sent. B. The digits are stripped off before matching the outbound POTS dial peer. C. The digits are stripped off by the connection trunk and R2 receives only D. R1 collects the 1200 and prepends the tie-line digits That number is matched to a VOIP dial peer and sent to the appropriate address. Correct Answer: A /Reference: QUESTION 45 Refer to exhibit. The Carmichael caller dials the site access code for Merrimack (6) by the four-digit extension number of the destination phone (0124). If the call is going to go accross the IP WAN, which action will have to be taken?

30 A. Translate to B. Strip the site access code and send four digits. C. Strip the site access 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, the prepend the access code when it reaches the Merrimack gateway. Correct Answer: B /Reference: QUESTION 46 Refer to the exhibit. You have a client that is testing a directory gatekeeper in the lab to provide address resolution between two different zones. Two of the commands in the running-config output are incorrect. Which two changes will correct the configuration? (Choose two.)

31 A. replace zone prefix GK-A with zone prefix GK-A B. replace zone prefix GK-B with zone prefix GK-B C. replace zone local DGK acme.com with zone remote DGK acme.com D. replace zone local GK-A acme.com with zone remote GK-A acme.com E. replace zone remote-gk-b acme.com with zone local GK-B acme.com

32 Correct Answer: AD /Reference: QUESTION 47 A. The voice packets are routed through the call agent. B. The voice packets are routed through the gatekeeper. C. The voice packets travel directly from phone to phone. D. The first call leg terminates at gateway A. The second call leg is from gateway A to phone G. E. The first call leg terminates at gateway B. The second call leg is from gateway B to phone G. F. The first call leg terminates at gateway A. The second call leg is from gateway A to its termination at gateway B. The third call leg is from gateway B to phone G. Correct Answer: C /Reference: QUESTION 48 Refer to the exhibit. Your customers dial in to your company using a local number, and their calls cross the WAN to an IVR system. They are complaining tha the IVR system does not always accept their group or may get it wrong. The IVR system has been checked and is working properly. What needs to be added to the dial peer on the incoming H.323 gateway to correct this problem?

33 A. no vad B. tech-prefix 1# C. codec g729ar8 bytes 30 D. dtmf-relay h245-alphanumeric Correct Answer: D /Reference: QUESTION 49 What is the best description of an MGCP endpoint? A. the interconnection between packet and traditional telephone networks B. any analog telephony device (PBX, switch, etc.) C. IP phones D. the gatekeepers in a VOIP network Correct Answer: A /Reference: QUESTION 50 In North America, which E&M signaling type is used most often for geographically separated equipment? A. Type I B. Type II C. Type III D. Type IV E. Type V Correct Answer: B /Reference: QUESTION 51 Which three are supervisory signals? (Choose Three.) A. busy

34 B. on hook C. off hook D. call waiting E. ring Correct Answer: BCE /Reference: QUESTION 52 A. Configure WRR with voice as highest priority. Use ACLs to classify voice traffic. Isolate voice traffic in its own VLAN. Configure access switches to trust traffic from IP phones. B. Configure a PQ with WRR. Use ACLs to classify voice control traffic. Isolate voice traffic in its own VLAN. Configure access switches to trust traffic from IP phones. C. Configure a PQ with WRR. Use ACLs to classify voice traffic. Isolate voice traffic in its own VLAN. Configure access switches to trust traffic from IP phones. D. Configure a PQ with voice control as highest priority. Use ACLs to classify voice control traffic. Isolate voice control traffic in its own VLAN. Configure access switches to trust voice control traffic from IP phones. Correct Answer: C /Reference: QUESTION 53 The D channel in ISDN is an example of which two signaling methods? (Choose two.) A. CAS

35 B. CCS C. In-band D. gateway E. out-of-band Correct Answer: BE /Reference: QUESTION 54 A. voice port 1/0/0 signaling wink-start operation 4-wire auto-cut-through type 1 B. voice port 1/0/0 signaling immediate-start operation 4-wire type5 C. voice port 1/0/0 signaling delay-start auto-cut-trhough operation 4-wire type 3 D. voice port 1/0/0 signaling wink-start operation 4-wire type 4 Correct Answer: A /Reference:

36 QUESTION 55 Which dial plan characteristic shows the most obvious improvement by dropping a number translation step? A. availability B. post-dial delay C. scalability D. hierarchical design Correct Answer: B /Reference: QUESTION 56 A. incorrect destination-pattern in router 1 B. incorrect POTS dial-peer statement in router 2 C. incorrect session-target statement in router 2 D. incorrect port statement in router 1 pots dial peer

37 E. missing no digit-strip on the voip dial peer in router 1 Correct Answer: A /Reference: QUESTION 57 What is the most common E&M type used outside North America? A. Type I B. Type II C. Type III D. Type IV E. Type V Correct Answer: E /Reference: QUESTION 58 Site A uses three-digit internal numbers and remote site 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. Correct Answer: C /Reference: QUESTION 59 Refer to the exhibit. Enzo's Bikes manufactures high end bicycle frames. Until recently they sold only to bicycle shops; however, now they are starting to sell to end users. They need a way to add two additional sales staff and ensure that the senior sales technician always gets the first call. Drew is the senior sales technician. Bob is the newest sales technician. Bob's phone should always be the last one chosen for incoming sales calls, after Drew and James. Bob's phone should be chosen first only when Drew and James are busy on calls. Select the correct dial-peer command set for Bob's phone.

38 A. dial-peer voice 3 pots destination-pattern preference 2 B. dial-peer voice 3 pots destination-pattern preference 0 C. dial-peer voice 3 pots destination-pattern preference firstlast D. dial-peer voice 3 pots destination-pattern preference 3 huntstop E. dial-peer voice 3 pots destination-pattern preference high Correct Answer: A /Reference: QUESTION 60

39 A. dial-peer voice 1 pots destination-pattern 5552.[0-5]0 B. dial-peer voice 1 pots destination pattern 5552[5-6].0 C. dial-peer voice 1 pots destination-pattern 555[2-5][5-6] D. dial-peer voice 1 pots destination-pattern 5552[5-6][05]0 Correct Answer: D /Reference: QUESTION 61 Select and Place:

40 Correct Answer:

41 /Reference: QUESTION 62 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 Correct Answer: D /Reference: QUESTION 63 Refer to the IOS configuration in the exhibit. How will the next incoming call be routed?

42 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 avaliable channel. D. The call will be routed to the next avaliable channel, starting from channel 1, hunting up toward channel 24. E. The call will be routed to the next avaliable channel, starting from channel 24, hunting down toward channel 1. Correct Answer: E /Reference: QUESTION 64 A new business in Great Britain needs to have a PSTN connection that will handle a maximum of 30 inbound and outbound calls at any given time. The customer only has one slot available on the designated PSTN router. Which digital line type should be recommended? A. QSIG B. ISDN BRI C. ISND E1 PRI D. ISDN T1 PRI Correct Answer: C /Reference: QUESTION 65

43 A. dial-peer voice 1374 pots destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip corlist incoming Intl01 destination-pattern 9011T session target ipv4: B. dial-peer voice 1374 pots destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip corlist outgoing Intl01 destination pattern 9011T C. dial-peer voice 1374 pots corlist incoming LDLst destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip destination-pattern 9011T session target ipv4: D. dial-peer voice 1374 pots corlist outgoing LDLst destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip destination-pattern 9011T

44 session target ipv4: E. dial-peer voice 1374 pots corlist incoming LocalLst destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip corlist outgoing Intl01 destination-pattern 9011T session target ipv4: F. dial-peer voice 1374 pots corlist outgoing LDLst destination-pattern 1374 port 1/0/0 dial-peer voice 100 voip corlist incoming Intl01 destination-pattern 9011T session target ipv4: Correct Answer: E /Reference:

642-436 Q&A. DEMO Version

642-436 Q&A. DEMO Version Cisco Voice over IP (CVOICE) Q&A DEMO Version Copyright (c) 2010 Chinatag LLC. All rights reserved. Important Note Please Read Carefully For demonstration purpose only, this free version Chinatag study

More information

CVOICE - Cisco Voice Over IP

CVOICE - Cisco Voice Over IP CVOICE - Cisco Voice Over IP Table of Contents Introduction Audience At Course Completion Prerequisites Applicable Products Program Contents Course Outline Introduction This five-day course covers the

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Course Objectives Explain the benefits and components of a Cisco Unified Communications system Describe how traditional telephony

More information

Voice Dial Plans, Configuring Voice Interfaces and Dial Peers

Voice Dial Plans, Configuring Voice Interfaces and Dial Peers Voice Dial Plans, Configuring Voice Interfaces and Dial Peers Cisco Networking Academy Program 1 Call Establishment Principles 2 Dial-Peer Call Legs 3 End-to-End Calls 4 Configuring Dial Peers 5 Understanding

More information

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX)

Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Implementing Cisco IOS Telephony and Unified Communications Express (IITUCX) Who should attend The primary audience for this course is as follows: Network administrators Network engineers Systems engineers

More information

Building a Scalable Numbering Plan

Building a Scalable Numbering Plan Building a Scalable Numbering Plan Scalable Numbering Plan This topic describes the need for a scalable numbering plan in a VoIP network. Dial Plans Dial plans contain specific dialing patterns for a user

More information

Dial Peer. Example: Dial-Peer Configuration

Dial Peer. Example: Dial-Peer Configuration Configuring Dial Peers Understanding Dial Peers This topic describes dial peers and their applications. Understanding Dial Peers A dial peer is an addressable call endpoint. Dial peers establish logical

More information

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX)

IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) Temario IMPLEMENTING CISCO IOS TELEPHONY AND UNIFIED COMMUNICATIONS EXPRESS (IITUCX) This course is designed to be the primary training for Cisco Unified Communications Manager Express and Cisco Unity

More information

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1

IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 IMPLEMENTING CISCO VOICE COMMUNICATIONS AND QOS Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional References Cisco Glossary of Terms Your Training

More information

Let's take a look at another example, which is based on the following diagram:

Let's take a look at another example, which is based on the following diagram: Chapter 3 - Voice Dial Peers In order to understand the concept of dial peers, it is important to understand call legs. A voice call over a packet network is segmented into discrete call legs. A call leg

More information

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program

IP Telephony v1.0 Scope and Sequence. Cisco Networking Academy Program IP Telephony v1.0 Scope and Sequence Cisco Networking Academy Program Table of Content COURSE OVERVIEW...4 Course Description...4 Course Objectives...4 Target Audience...5 Prerequisites...5 Lab Requirements...5

More information

640-460 - Implementing Cisco IOS Unified Communications (IIUC)

640-460 - Implementing Cisco IOS Unified Communications (IIUC) 640-460 - Implementing Cisco IOS Unified Communications (IIUC) Course Introduction Course Introduction Module 1 - Cisco Unified Communications System Introduction Cisco Unified Communications System Introduction

More information

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>>

640-460. IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo. Page <<1/9>> 640-460 IIUC Implementing Cisco IOS Unified Communications (IIUC) Version: Demo Page 1. You are CCNA VOICE associate in XXXX.com. You need configure a voice port that will allow the gateway to

More information

Implementing Cisco Voice Communications and QoS

Implementing Cisco Voice Communications and QoS Implementing Cisco Voice Communications and QoS Course CVOICE v8.0; 5 Days, Instructor-led Course Description Implementing Cisco Voice Communications and QoS (CVOICE) v8.0 teaches learners about voice

More information

Call Setup and Digit Manipulation

Call Setup and Digit Manipulation Call Setup and Digit Manipulation End-to-End Calls This topic explains how routers interpret call legs to establish end-to-end calls. End-to-End Calls IP Telephony 2005 Cisco Systems, Inc. All rights reserved.

More information

Configuring Network Side ISDN PRI Signaling, Trunking, and Switching

Configuring Network Side ISDN PRI Signaling, Trunking, and Switching Configuring Network Side ISDN PRI Signaling, Trunking, and Switching This chapter describes the Network Side ISDN PRI Signaling, Trunking, and Switching feature. The following main sections are provided:

More information

This topic describes dial peers and their applications.

This topic describes dial peers and their applications. Dial Peers What is Dial Peer? This topic describes dial peers and their applications. What is a Dial Peer? A dial peer is an addressable call endpoint. Dial peers establish logical connections, called

More information

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005

CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 CVOICE Exam Topics Cisco Voice over IP Exam #642-432 7/14/2005 From the Cisco CVOICE 642-432 Exam Topics Voice over IP Technologies Describe the similarities and differences between PSTN and VoIP including

More information

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved.

Cisco Networks (ONT) 2006 Cisco Systems, Inc. All rights reserved. Optimizing Converged Cisco Networks (ONT) reserved. Lesson 2.4: Calculating Bandwidth Requirements for VoIP reserved. Objectives Describe factors influencing encapsulation overhead and bandwidth requirements

More information

Introducing Cisco Voice and Unified Communications Administration Volume 1

Introducing Cisco Voice and Unified Communications Administration Volume 1 Introducing Cisco Voice and Unified Communications Administration Volume 1 Course Introduction Overview Learner Skills and Knowledge Course Goal and Course Flow Additional Cisco Glossary of Terms Your

More information

Cisco Voice Gateways. PacNOG6 VoIP Workshop Nadi, Fiji. November 2009. Jonny Martin - jonny@jonnynet.net

Cisco Voice Gateways. PacNOG6 VoIP Workshop Nadi, Fiji. November 2009. Jonny Martin - jonny@jonnynet.net Cisco Voice Gateways PacNOG6 VoIP Workshop Nadi, Fiji. November 2009 Jonny Martin - jonny@jonnynet.net Voice Gateways Any device with one or more TDM PSTN interfaces on them TDM - Time Division Multiplexing

More information

Optimizing Converged Cisco Networks (ONT)

Optimizing Converged Cisco Networks (ONT) Optimizing Converged Cisco Networks (ONT) Module 2: Cisco VoIP Implementations (Deploy) Calculating Bandwidth Requirements for VoIP Objectives Describe factors influencing encapsulation overhead and bandwidth

More information

642-437. Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Version: Demo. Page <<1/8>>

642-437. Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Version: Demo. Page <<1/8>> 642-437 Implementing Cisco Unified Communications Voice over IP and QoS v8.0 (CVOICE v8.0) Version: Demo Page 1. Which three Cisco IOS commands are required to configure a voice gateway as a DHCP

More information

VoIP Signaling and Call Control

VoIP Signaling and Call Control VoIP Signaling and Call Control Cisco Networking Academy Program 1 Need for Signaling and Call Control 2 Model for VoIP Signaling and Call Control VoIP signaling components Endpoints Common control Common

More information

Cisco CME SIP Trunk Configuration

Cisco CME SIP Trunk Configuration Cisco CME SIP Trunk Configuration There are lots of example configurations on the Internet that illustrate how to connect CME to SIP trunks. Few offered any insight as to the reason for the commands that

More information

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports

Table of Contents. Cisco Mapping Outbound VoIP Calls to Specific Digital Voice Ports Table of Contents Mapping Outbound VoIP Calls to Specific Digital Voice Ports...1 Introduction...1 Before You Begin...1 Conventions...1 Prerequisites...1 Components Used...1 Configure...2 Network Diagram...2

More information

Cisco ISDN PRI to SIP Gateway

Cisco ISDN PRI to SIP Gateway Cisco ISDN PRI to SIP Gateway Supported features Full ISDN E1 emulation Early media support Inbound calling. Type SIP REGISTERED TRUNK Outbound Calling ISDN PRI equivalent Secure Calling via SIP Encrypt

More information

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course

Version dated 25/11/2014. 1.Course Title. NATO Voice over IP Foundation Course. 2.Identification Number (ID) 3. Purpose of the Course 1.Course Title Version dated 25/11/2014 NATO Voice over IP Foundation Course 2.Identification Number (ID) 095 3. Purpose of the Course There are a number of new technologies (to NATO) that are encompassed

More information

Curso de Telefonía IP para el MTC. Sesión 5-1 Implementación de Gateways SIP. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 5-1 Implementación de Gateways SIP. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 5-1 Implementación de Gateways SIP Mg. Antonio Ocampo Zúñiga SIP Fundamentals SIP is a simple extensible protocol. SIP is defined in IETF RFC 2543 and RFC 3261.

More information

- Basic Voice over IP -

- Basic Voice over IP - 1 Voice over IP (VoIP) - Basic Voice over IP - Voice over IP (VoIP) is a digital form of transport for voice transmissions, replacing analog phone systems. The benefits of VoIP are considerable: Better

More information

Network Scenarios Pagina 1 di 35

Network Scenarios Pagina 1 di 35 Network Scenarios Pagina 1 di 35 Table of Contents Network Scenarios Cisco 827 s Network Connections Internet Access Scenarios Before You Configure Your Internet Access Network Replacing a Bridge or Modem

More information

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy

AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy INTERACTIVE INTELLIGENCE AT&T IP Flex Reach/ IP Toll Free Configuration Guide IC 3.0 with Interaction SIP Proxy Version 1.7 9/2/2009 TABLE OF CONTENTS 1 AT&T... 5 1.1 Introduction... 5 1.2 Product Descriptions...

More information

White paper. SIP An introduction

White paper. SIP An introduction White paper An introduction Table of contents 1 Introducing 3 2 How does it work? 3 3 Inside a normal call 4 4 DTMF sending commands in sip calls 6 5 Complex environments and higher security 6 6 Summary

More information

Special-Purpose Connections

Special-Purpose Connections Special-Purpose Connections Connection Commands This topic identifies different special-purpose connection commands. Special-Purpose Connection Commands connection plar Associates a voice port directly

More information

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction

640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction 640-461: Introducing Cisco Voice and Unified Communications Administration (ICOMM) v8.0 Course Introduction Course Introduction Module 01 - Overview of Cisco Unified Communications Solutions Understanding

More information

Operation Manual Voice Overview (Voice Volume) Table of Contents

Operation Manual Voice Overview (Voice Volume) Table of Contents Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over... 1-1 1.1 Introduction to VoIP... 1-1 1.1.1 VoIP System... 1-1 1.1.2 Basic VoIP Call Flow... 1-2 1.1.3

More information

Dial Peer Configuration Examples

Dial Peer Configuration Examples Dial Peer Configuration Examples This appendix contains a series of configuration examples featuring the minimum required components and critical Cisco IOS command lines extracted from voice gateway configuration

More information

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1

Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Cablevision Optimum Voice SIP Trunking with Avaya IP Office - Issue 1.1 Abstract These Application Notes describe the procedures

More information

Integrating VoIP Phones and IP PBX s with VidyoGateway

Integrating VoIP Phones and IP PBX s with VidyoGateway Integrating VoIP Phones and IP PBX s with VidyoGateway Updated February 2011 INDEX: I. ABSTRACT.1 II. III. IV. VIDYOGATEWAY OVERVIEW.. 1 NETWORK TOPOLOGIES AND DEFINITIONS...2 CONNECTING TO VIDYOCONFERENCES

More information

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 1 Introducción. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 1 Introducción Mg. Antonio Ocampo Zúñiga Conceptos Generales VoIP Essentials Family of technologies Carries voice calls over an IP network VoIP services convert

More information

IP Telephony Deployment Models

IP Telephony Deployment Models CHAPTER 2 Sections in this chapter address the following topics: Single Site, page 2-1 Multisite Implementation with Distributed Call Processing, page 2-3 Design Considerations for Section 508 Conformance,

More information

Need for Signaling and Call Control

Need for Signaling and Call Control Need for Signaling and Call Control VoIP Signaling In a traditional voice network, call establishment, progress, and termination are managed by interpreting and propagating signals. Transporting voice

More information

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction

Hands on VoIP. Content. Tel +44 (0) 845 057 0176 enquiries@protelsolutions.co.uk. Introduction Introduction This 4-day course offers a practical introduction to 'hands on' VoIP engineering. Voice over IP promises to reduce your telephony costs and provides unique opportunities for integrating voice

More information

Session Initiation Protocol (SIP) The Emerging System in IP Telephony

Session Initiation Protocol (SIP) The Emerging System in IP Telephony Session Initiation Protocol (SIP) The Emerging System in IP Telephony Introduction Session Initiation Protocol (SIP) is an application layer control protocol that can establish, modify and terminate multimedia

More information

Gateways and Their Roles

Gateways and Their Roles Gateways and Their Roles Understanding Gateways This topic describes the role of voice gateways and their application when connecting VoIP to traditional PSTN and telephony equipment. Analog vs. Digital

More information

Mediatrix 3000 with Asterisk June 22, 2011

Mediatrix 3000 with Asterisk June 22, 2011 Mediatrix 3000 with Asterisk June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 Network Topology... 3 Equipment Detail... 3 Configuration of the Fax Extension... 4 Configuration

More information

Troubleshooting Voice Over IP with WireShark

Troubleshooting Voice Over IP with WireShark Hands-On Course Description Voice over IP is being widely implemented both within companies and across the Internet. The key problems with IP voice services are maintaining the quality of the voice service

More information

Cisco Voice over IP CIPT1. Volume 2. Student Guide. Version 6.0. Editorial, Production, and Web Services: 02.15.08

Cisco Voice over IP CIPT1. Volume 2. Student Guide. Version 6.0. Editorial, Production, and Web Services: 02.15.08 CIPT1 Cisco Voice over IP Volume 2 Version 6.0 Student Guide Editorial, Production, and Web Services: 02.15.08 DISCLAIMER WARRANTY: THIS CONTENT IS BEING PROVIDED AS IS. CISCO MAKES AND YOU RECEIVE NO

More information

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0

Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Avaya IP Office 9.0 with HIPCOM SIP Trunk Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Configuring Voice over IP

Configuring Voice over IP CHAPTER 4 This chapter explains how to configure voice interfaces and ports, which convert telephone voice signals for transmission over an IP network. This chapter presents the following major topics:

More information

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week

IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week Course Title: No. of Hours: IP Implementation in Private Branch Exchanges From 9:30 a.m until 4:30 p.m (7 hrs./day) 5 days / week 1 Course Duration: 3 Months (12weeks) No. Of Hours: 7 Hrs./Day- 5 days/week.

More information

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0

Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Avaya IP Office 7.0 Integration with Skype Connect R2.0 Issue 1.0 Abstract These Application Notes describe the steps to configure an Avaya

More information

SIP Trunking and Voice over IP

SIP Trunking and Voice over IP SIP Trunking and Voice over IP Agenda What is SIP Trunking? SIP Signaling How is Voice encoded and transported? What are the Voice over IP Impairments? How is Voice Quality measured? VoIP Technology Confidential

More information

Cisco IOS SIP Configuration Guide

Cisco IOS SIP Configuration Guide Cisco IOS SIP Configuration Guide Dialpeer Configuration Session Number 1 Terminology Call - A connection terminating on or passing through a gateway. Call Leg - The segment of a call associated with a

More information

Avaya IP Office 8.1 Configuration Guide

Avaya IP Office 8.1 Configuration Guide Avaya IP Office 8.1 Configuration Guide Performed By tekvizion PVS, Inc. Contact: 214-242-5900 www.tekvizion.com Revision: 1.1 Date: 10/14/2013 Copyright 2013 by tekvizion PVS, Inc. All Rights Reserved.

More information

Frequently Asked Questions about Integrated Access

Frequently Asked Questions about Integrated Access Frequently Asked Questions about Integrated Access Phone Service How are local, long distance, and international calls defined? Local access transport areas (LATAs) are geographical boundaries set by the

More information

H.323 and Associated Recommendations. This topic describes H.323 and its protocols and explains how H.323 is used in the IP internetwork environment.

H.323 and Associated Recommendations. This topic describes H.323 and its protocols and explains how H.323 is used in the IP internetwork environment. Configuring H.323 H.323 and Associated Recommendations This topic describes H.323 and its protocols and explains how H.323 is used in the IP internetwork environment. H.323 and Associated Recommendations

More information

HOWTO on Virtual Voice Rack

HOWTO on Virtual Voice Rack HOWTO on Virtual Voice Rack Petr Lapukhov CCIE #16379 (R&S/Security/SP) InternetworkExpert Inc. Motivation One day it happens. You decide to go after your Voice CCIE, and nothing can stop you. It takes

More information

NTP VoIP Platform: A SIP VoIP Platform and Its Services

NTP VoIP Platform: A SIP VoIP Platform and Its Services NTP VoIP Platform: A SIP VoIP Platform and Its Services Speaker: Dr. Chai-Hien Gan National Chiao Tung University, Taiwan Email: chgan@csie.nctu.edu.tw Date: 2006/05/02 1 Outline Introduction NTP VoIP

More information

EarthLink Business SIP Trunking. Cisco Call Manager and Cisco CUBE Customer Configuration Guide

EarthLink Business SIP Trunking. Cisco Call Manager and Cisco CUBE Customer Configuration Guide EarthLink Business SIP Trunking Cisco Call Manager and Cisco CUBE Customer Configuration Guide Publication History First Release: Version 2.0 April 20, 2012 CHANGE HISTORY Version Date Change Details Changed

More information

Connecting Multiple Cisco Unified CallManager Express Systems with VoIP

Connecting Multiple Cisco Unified CallManager Express Systems with VoIP CHAPTER 6 Connecting Multiple Cisco Unified CallManager Express Systems with VoIP This chapter describes the ways in which you can use Cisco Unified CallManager Express (Cisco Unified CME) as a component

More information

Internet Telephony PBX System

Internet Telephony PBX System Internet Telephony PBX System T1/E1 Gateway With IP PBX Application Copyright PLANET Technology Corporation. All rights reserved. Case 35: With IP PBX Application Head Office E1 PABX interconnect with

More information

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Mediatrix 4404 Step by Step Configuration Guide June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents First Steps... 3 Identifying your MAC Address... 3 Identifying your Dynamic IP Address...

More information

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks)

2- Technical Training (9 weeks) 3- Applied Project (3 weeks) 4- On Job Training (OJT) (4 weeks) Course Title: Prerequisites: Training Program (5 months) IP Implementation in Private Branch Exchanges Must fresh graduates Communication/Electronics Engineers" 1- Soft Skills Training (4 weeks) 1. Communication

More information

SIP Trunking. Cisco Press. Christina Hattingh Darryl Sladden ATM Zakaria Swapan. 800 East 96th Street Indianapolis, IN 46240

SIP Trunking. Cisco Press. Christina Hattingh Darryl Sladden ATM Zakaria Swapan. 800 East 96th Street Indianapolis, IN 46240 SIP Trunking Christina Hattingh Darryl Sladden ATM Zakaria Swapan Cisco Press 800 East 96th Street Indianapolis, IN 46240 SIP Trunking Contents Introduction xix Part I: From TDM Trunking to SIP Trunking

More information

MAXCS Release 7.0. Application Note: Remote MultiVoIP Gateway Configuration. Intended audience: AltiGen Authorized Partners

MAXCS Release 7.0. Application Note: Remote MultiVoIP Gateway Configuration. Intended audience: AltiGen Authorized Partners MAXCS Release 7.0 Application Note: Remote MultiVoIP Gateway Configuration Intended audience: AltiGen Authorized Partners March 4, 2014 Contents Introduction... 3 Requirements... 3 MultiVoIP Gateway Configuration...

More information

SIP Trunking Manual 05.15. Technical Support Web Site: http://ws1.necii.com (registration is required)

SIP Trunking Manual 05.15. Technical Support Web Site: http://ws1.necii.com (registration is required) SIP Trunking Manual 05.15 Technical Support Web Site: http://ws1.necii.com (registration is required) This manual has been developed by NEC Unified Solutions, Inc. It is intended for the use of its customers

More information

VoIP Configuration Examples

VoIP Configuration Examples APPENDIX C This section uses four different scenarios to demonstrate how to configure Voice over IP (VoIP). The actual VoIP configuration procedure depends on the topology of your voice network. The following

More information

nexvortex Setup Guide

nexvortex Setup Guide nexvortex Setup Guide CISCO UC500 March 2012 Introduction This document is intended only for nexvortex customers and resellers as an aid to setting up the Cisco PBX software to connect to the nexvortex

More information

ESI SIP Trunking Installation Guide

ESI SIP Trunking Installation Guide ESI SIP Trunking Installation Guide 0450-1227 Rev. B Copyright 2009 ESI (Estech Systems, Inc.). Information contained herein is subject to change without notice. ESI products are protected by various U.S.

More information

Application Notes Rev. 1.0 Last Updated: February 3, 2015

Application Notes Rev. 1.0 Last Updated: February 3, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v8.6 for Level 3 Voice Complete SM Deployments Application Notes Rev. 1.0 Last Updated: February 3, 2015 Contents 1 Document Overview...

More information

GW400 VoIP Gateway. User s Guide

GW400 VoIP Gateway. User s Guide GW400 VoIP Gateway User s Guide P/N: 956YD30001 Copyright 2006. All Rights Reserved. Document Version: 1.0 All trademarks and trade names are the properties of their respective owners. i Table of Contents

More information

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2

Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Network Connection Considerations for Microsoft Response Point 1.0 Service Pack 2 Updated: February 2009 Microsoft Response Point is a small-business phone solution that is designed to be easy to use and

More information

Configuration Notes 0217

Configuration Notes 0217 PBX Remote Line Extension using Mediatrix 1104 and 1204 Introduction... 2 Application Scenario... 2 Running the Unit Manager Network (UMN) Software... 3 Configuring the Mediatrix 1104... 6 Configuring

More information

Curso de Telefonía IP para el MTC. Sesión 4-1 Tipos de llamadas. Mg. Antonio Ocampo Zúñiga

Curso de Telefonía IP para el MTC. Sesión 4-1 Tipos de llamadas. Mg. Antonio Ocampo Zúñiga Curso de Telefonía IP para el MTC Sesión 4-1 Tipos de llamadas Mg. Antonio Ocampo Zúñiga Call Types Local: Does not traverse the WAN or PSTN. On-net: Occurs between two telephones on the same data network.

More information

How To Set Up A Dialogic.Com On A Cell Phone With A Sim Sim Sims On A Sims 2 (For A Simplon) On A Pts 2 ( For A Pty Phone) On An Ipad Or

How To Set Up A Dialogic.Com On A Cell Phone With A Sim Sim Sims On A Sims 2 (For A Simplon) On A Pts 2 ( For A Pty Phone) On An Ipad Or Dialogic Brooktrout SR140 Fax Software with Cisco Unified Communications Manager 8.0 IMPORTANT NOTE This document is not to be shared with or disseminated to other third parties, in whole or in part, without

More information

Cisco CCA Tool SIP Security methods

Cisco CCA Tool SIP Security methods Cisco CCA Tool SIP Security methods The Cisco CCA tool (Cisco Configuration Assistant) provides a graphical interface for configuring the UC500 series devices. Once settings have been established using

More information

Configuring Voice over IP

Configuring Voice over IP Configuring Voice over IP This chapter provides an overview of Voice over IP (VoIP) technology and gives step-by-step configuration tasks. The chapter contains the following sections: VoIP Benefits, page

More information

NAT TCP SIP ALG Support

NAT TCP SIP ALG Support The feature allows embedded messages of the Session Initiation Protocol (SIP) passing through a device that is configured with Network Address Translation (NAT) to be translated and encoded back to the

More information

Table of Contents. Confidential and Proprietary

Table of Contents. Confidential and Proprietary Table of Contents About Toshiba Strata CIX and Broadvox SIP Trunking... 1 Requirements... 2 Purpose, Scope and Audience... 3 What is SIP Trunking?... 4 Business Advantages of SIP Trunking... 4 Technical

More information

NetVanta 7100 Exercise Service Provider SIP Trunk

NetVanta 7100 Exercise Service Provider SIP Trunk NetVanta 7100 Exercise Service Provider SIP Trunk PSTN NetVanta 7100 FXS 0/1 x2001 SIP Eth 0/0 x2004 SIP Server 172.23.102.87 Hosted by x2003 www.voxitas.com In this exercise, you will create a SIP trunk

More information

VoIP Gateway/IP-PBX Interworking with Skype

VoIP Gateway/IP-PBX Interworking with Skype VoIP Gateway/IP-PBX Interworking with Skype IPNext50 IP-PBX IPNext20 IP-PBX AP100B VoIP Gateway AddPac Technology 2010, Sales and Marketing www.addpac.com Contents Skype Interworking Test VoIP Gateway

More information

IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution

IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution IPNext 50 NGN IP-PBX High-performance Next Generation IP-PBX Solution IP-PBX Features www.addpac.com AddPac Technology 2008, Sales and Marketing Contents IP-PBX Features Smart Multimedia Manager VoIP Gateway

More information

Application Notes Rev. 1.0 Last Updated: January 9, 2015

Application Notes Rev. 1.0 Last Updated: January 9, 2015 SBC 1000/2000 Series Configuration Guide with Cisco Unified Call Manager v9.1 for Level 3 Voice Complete SM SIP Trunk Deployments Application Notes Rev. 1.0 Last Updated: January 9, 2015 Contents 1 Document

More information

Cisco Unified Video Conferencing Configuration

Cisco Unified Video Conferencing Configuration Cisco Unified Video Conferencing Configuration This topic provides a reference configuration for Cisco Unified Video Conferencing within a Cisco Unified Communications deployment. The information is based

More information

ICANWK610A Design and build integrated VoIP networks

ICANWK610A Design and build integrated VoIP networks ICANWK610A Design and build integrated VoIP networks Release: 1 ICANWK610A Design and build integrated VoIP networks Modification History Release Release 1 Comments This Unit first released with ICA11

More information

Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670

Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Configuration of Applied VoIP Sip Trunks with the Toshiba CIX40, 100, 200 and 670 Businesses Save Money with Toshiba s New SIP Trunking Feature Unlike gateway based solutions, Toshiba s MIPU/ GIPU8 card

More information

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0

Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for BT Wholesale/HIPCOM SIP Trunk Service and Avaya IP Office 8.0 Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Development of SIP-H.323 Gateway Project

Development of SIP-H.323 Gateway Project Development of SIP-H.323 Gateway Project Ruston Hutchens QUESTnet 2005 Thursday 7 th July v2 SIP-H.323 Gateway project Motivation Large deployment base of H.323 terminals (over 2.9 million calls placed

More information

Avaya one-x Quick Edition Interoperability with Cisco Integrated Services Router (ISR) SIP Gateway - Issue 1.0

Avaya one-x Quick Edition Interoperability with Cisco Integrated Services Router (ISR) SIP Gateway - Issue 1.0 Avaya Solution & Interoperability Test Lab Avaya one-x Quick Edition Interoperability with Cisco Integrated Services Router (ISR) SIP Gateway - Issue 1.0 Abstract Avaya one-x Quick Edition R3.1 provides

More information

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide

EarthLink Business SIP Trunking. NEC SV8100 IP PBX Customer Configuration Guide EarthLink Business SIP Trunking NEC SV8100 IP PBX Customer Configuration Guide Publication History First Release: Version 1.0 August 30, 2011 CHANGE HISTORY Version Date Change Details Changed By 1.0 8/30/2011

More information

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online 1 IP PBX SD Card Slot FXO Ports PBX LAN port PBX WAN port FXO Ports LED, RED means online 2 Connect the IP PBX to Your LAN Internet PSTN Router Ethernet Switch FXO Ports 3 Access the PBX s WEB GUI The

More information

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0

Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Intelepeer SIP Trunking with Avaya IP Office 7.0 - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information

Configuration Notes 290

Configuration Notes 290 Configuring Mediatrix 41xx FXS Gateway with the Asterisk IP PBX System June 22, 2011 Proprietary 2011 Media5 Corporation Table of Contents Introduction... 3 About Mediatrix 41xx Series FXS Gateways...

More information

IP Office Technical Tip

IP Office Technical Tip IP Office Technical Tip Tip no: 188 Release Date: September 27, 2007 Region: GLOBAL Verifying IP Office SIP Trunk Operation IP Office back-to-back SIP Line testing IP Office Release 4.0 supports SIP trunking.

More information

Cisco Multiservice IP-to-IP Gateway the Cisco IOS Session Border Controller

Cisco Multiservice IP-to-IP Gateway the Cisco IOS Session Border Controller Cisco Multiservice IP-to-IP Gateway the Cisco IOS Session Border Controller Cisco Unified Communications is a comprehensive IP communications system of voice, video, data, and mobility products and applications.

More information

Note: As of Feb 25, 2010 Priority Telecom has not completed FXS verification of fax capabilities. This will be updated as soon as verified.

Note: As of Feb 25, 2010 Priority Telecom has not completed FXS verification of fax capabilities. This will be updated as soon as verified. 1 SIP Carriers 1.1 Priority Telecom 1.1.1 Warnings Check the SIP 3 rd Party SIP Carrier Matrix for certification status, and supported features. More info about the SIP 3 rd Party SIP Carrier Matrix can

More information

Integrate VoIP with your existing network

Integrate VoIP with your existing network Integrate VoIP with your existing network As organisations increasingly recognise and require the benefits voice over Internet Protocol (VoIP) offers, they stop asking "Why?" and start asking "How?". A

More information

SIP Trunking Quick Reference Document

SIP Trunking Quick Reference Document SIP Trunking Quick Reference Document Publication Information SAMSUNG TELECOMMUNICATIONS AMERICA reserves the right without prior notice to revise information in this publication for any reason. SAMSUNG

More information

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0

Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Avaya Solution & Interoperability Test Lab Application Notes for Configuring Broadvox SIP Trunking with Avaya IP Office - Issue 1.0 Abstract These Application Notes describe the procedures for configuring

More information