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Found 24 results

  1. hi all, please i need to answer the below question and thanks in advance .
  2. HI Guys, I see INE has released a bunch of CCIE Collab V3 videos series....I will very grateful if someone can share the below courses: Introduction to CCIE Collab V3 [Hidden Content] Cisco Unified Contact Center Express (UCCX) - [Hidden Content] Unity Connection - [Hidden Content] Cisco Unified Communications IM and Presence (IM&P) - [Hidden Content] Cisco Meeting Server (CMS) - [Hidden Content] Expressway - [Hidden Content] Intercluster Lookup Service (ILS) - [Hidden Content] Cisco Unified Border Element (CUBE) - [Hidden Content] Quality of Service (QoS) - [Hidden Content] Thanks in advance!!
  3. can anyone share latest version of CIPC for windows?
  4. Please share CCNP Collaboration CLCOR & CLACCM - Cisco Netacad PPTX & Labs
  5. Dears Please can some one share the Khawar Butt CCIE Collaboration videos. Thanks and Best Regards
  6. Hi Guys, So am following this CCNA Voice lab whihc is pretty good. Its just at 32:15min when he showing the ephone-dn 1 part i am getting an error (see below) R1(config)# *Mar 1 01:01:05.403: %DIALPEER_DB-3-ADDPEER_MEM_THRESHOLD: Addition of dial-peers limited by available memory R1(config)# *Mar 1 01:01:05.407: %SYS-2-MALLOCFAIL: Memory allocation of 65536 bytes failed from 0x6003ACE0, alignment 8 Pool: Processor Free: 51452 Cause: Not enough free memory Alternate Pool: None Free: 0 Cause: No Alternate pool -Process= "Chunk Manager", ipl= 3, pid= 1 -Traceback= 0x61467204 0x60014798 0x6001A04C 0x6003ACE8 0x60039D94 0x60039B78 *Mar 1 01:01:05.415: %SYS-2-CHUNKEXPANDFAIL: Could not expand chunk pool for VTSP EVENT poo. No memory available -Process= "Chunk Manager", ipl= 3, pid= 1 -Traceback= 0x61467204 0x60039BBC I believe its memory issue but i cant see to fix it. Can anyone help?? thanks
  7. If anyone need the study material, i can share it.
  8. Hello, Can anyone share IPEXPERT' S CCIE Collab technology wb vol 1 or mock labs? Thanks Much
  9. Hi All, I'm in the midst of preparing the lab. and wanted to know is there anyone try on this workbook (ccxx collaboration workbook)? I want to know how accuratecy of this, the price also quite competitive compare with Cciecollborationlab.. TIA
  10. Have a lot of equipment, enough for four ccie collaboration racks. A lot of people ask me to build rack rental for collaboration so they can login remotely and practice. I would like to build rack rental just like INE has it but don't have an idea on how to start. Yes, I know how to setup the rack but I don't know how to interface with some kind of control panel where time will be log for the user and gives some automation to the user to save and open their config for the session. If anyone can point me in the right direction I would really appreciate it.
  11. Hi In our usual manner, if there's anyone who is interested in CCIE COLLABORATION like me, can we use this thread to discuss it? We can share useful materials, knowledge, group buy etc. I will also like to have a skype group out of this if this is successful. Let me know if this is in your direction and share how you plan to help others through this thread. If we join force. You can PM me if you think that is more appropriate. Do not Post Skype IDs below or else face ban. Some useful stuff: 1. ccie collaboration quick reference guide: [Hidden Content] 2. CollaborationCcisco live: [Hidden Content] 3. Kevin Wallace voice stuff: [Hidden Content] 4. Cisco UCCX Scripting Video: [Hidden Content] 5. SIP Trunking [Hidden Content] ****** Lab Question sample****** Note: Lab Diagram is omitted right now. PSTN Dialing Instructions: The PSTN Phone is used place/receive calls to/from each of your sites. When calling from PSTN Phone to any of your site DID number, please remember the following rules. 1. You do not need to dial any leading Digit. 2. Dialing from PSTN Phone to Site DID number is localized. Please see point 3,4& 5 below. 3. Line 1 and 2 assumes US dialing patterns 1. 7 digits for local calls 2. 1 + <10 digits> for long distance calls only 3. 011 + <country code> + <any number of digits> for international calls 4. Line 4 and 6 assumes Hong Kong and U.K dialing patterns 1. 8 digits for local calls 2. 00 + <country code> + <any number of digits> for international calls 5. For example, to call, HQ Phone 1 (+14082022001) from PSTN, you need to dial 1. 2022001 if you use line 1, since line 1 belongs to local PSTN at HQ 2. 14082022001 if you use line 2, since line 2 is placing a US long distance call to reach the HQ area code 3. 0014082022001if you use line 4 or line 6, since these 2 lines is placing a international call to U.S. 6. Similarly, to call, SC Phone 1 from PSTN, you need to dial 1. 01185224044001 if you use line 1, since line 1 or 2 (U.S. PSTN) 2. 24044001 if you were dialing from line 4 (Local Hong Kong PSTN) 3. 0085224044001 if you were dialing from line 6 (U.K. PSTN) Section 1: CONFIGURE & TROUBLESHOOT CISCO COLLABORATION INFRASTRUCTURE 1.1 Voice and Data VLANs Configure Voice VLANs for your IP Phones at the HQ, SiteB and Site C. The voice VLAN numbers are 102 for HQ, 302 for Site B and 502 for Site C. Please refer to Topology Diagram and the Port Assignment tables for detailed information. There will be also PCs connected to the PC port of these IP phones. Configure the switch ports to place these PCs into the data VLAN, which are 202 for HQ, 402 for Site B and 602 for Site C. Score: 2 points 1.2 DHCP & NTP Services Configure the HQ Cisco Unified Communications Manager Subscriber server at 142.100.64.12 as the DHCP server to assign IP addresses to IP phones at the HQ Site. HQ IP phones need IP addresses from the local voice subnet (142.102.64.X/24) ranging from 142.102.64.30 to 142.102.64.50. Configure the Site B Cisco Unified Communications Manager Publisher server at 142.100.65.11 as the DHCP server to assign IP addresses to IP phones at the Site B. Assign addresses from the local voice subnet (142.102.65.X/24) in the range of 142.102.65.30 to 142.102.65.50 to Site B phones. Configure the Site C 2921 router (R3) as a Cisco IOS DHCP server, which assigns IP addresses to the IP phones at the local voice VLAN. IP phones at this site need IP addresses from the local voice subnet (142.102.66.X/24) ranging from 142.102.66.30 to 142.102.66.50. Also configure HQ CUCMs, Site B CUCM, R1, R2 and R3 to synchronize local clock with the backbone NTP server at 157.26.1.250. The backbone NTP server is running on Coordinated Universal Time. Score: 3 points Section 2: Configure and Troubleshoot Cisco Unified Communications Manager 2.1 CUCM IP Phones 2.1a- CUCM SCCP IP PHONE Register HQ phone 1 to HQ CUCM cluster and Site B phone 1 to Site B CUCM, according to provided telephony number scheme. Ensure the HQ phone is provisioned with TFTP redundancy such that the publisher CUCM is the backup TFTP server. Extension-to- extension dialing of each site uses 4 digits only. Caller name display should be delivered on internal 4- digit calls- use trivial caller names such as “HQ Phone 1” and “Site B Phone 1”. Lastly, both phones should display globalized calling number on the upper right hand corner of the phone screen. Refer to the exhibit below for an example captured on HQ Phone 1. .Score : 2 Points 2.1b-CUCM SIP IP PHONE Register and ConfigureHQ Phone 2 as SIP phone to the HQ Cluster. Ensure the phone is provisioned with TFTP redundancy such that the publisher CUCM is the backup TFTP server. Caller name display should be delivered on internal 4-digit calls- use trivial caller names such as “HQ phone 2”. Enable the video camera on HQ Phone 2. Globalized calling number should be displayed on the top of the phone screen. Refer to the exhibit below for an example captured as HQ Phone 2. Score: 2 Points 2.2-CUCM Gateways Information below is relevant to your digital T1 gateways. • Line coding/framing b8zfs/esf • ISDN switch type Primary- NI • Configure your gateways to take clock (layer1) from network. For PRI circuits configure your gateway as Layer 2 User side. • Refer to the individual gateway sub-sections for digit sending and receiving details. • Calling names should always be sent to the PSTN • Use full span on T1 PRI unless specified otherwise. • Gateway configuration must be verified with successful completion of inbound and outbound calls. Therefore, simple route pattern configuration is expected, even though call routing is not the key focus of this section. 2.2a HQ CUCM MGCP Gateway Configure and register R1 as IOS T1 PRI MGCP gateway to the HQ CUCM cluster. All MGCP traffic should use R1 local voice VLAN interface. Implement CUCM redundancy such that if the primary call agent CUCM subscriber goes down the back up call agent CUCM publisher Should be active. Telco delivers 10 digit direct inward dialing called numbers in inbound calls, 408202xxxx, where xxxx is any HQ internal 4-digits extension. Verify you HQ MGCP gateway configuration by placing an inbound call from line 1 on the PSTN phone to 202xxxx, where xxxx is any HQ internal 4-digit extension. For outbound calls, ensure 911 calls from any HQ IP phone terminate on R1. For 911 calls, send 408202xxxx, where xxxx is the 4-digit extension as the calling number. 9911 is not required. Calls traversing between HQ phone and R1 PRI trunk should always use G711 codec. Also ensure the following verification command is produced on R1. R1 = show ccm host MGCP domain name R1..com PRIMARY ---- 142.100.64.12 BACKUP ----- 142.100.64.11 Score: 2 Points 2.2b-SiteB CUCM MGCP Gateway Configure and register R2 as IOS T1 PRI MGCP gateway to the SiteB CUCM cluster. All MGCP traffic should use R2 local voice VLAN interfaces. Telco delivers 10-digits direct inward dialing (DID) called numbers on inbound calls 972303xxxx, where xxxx is any SiteB internal 4-digits extension. Verify your SiteB gateway configuration by placing an inbound call from PSTN phone line 2 to 303xxxx, where xxxx is any SiteB internal 4-digit extension. For outbound calls, ensure any SiteB IP phone call 911 through R2. For 911 calls, send 972303xxxx, where xxxx is the 4-digit extension, as the calling number. You are not required to configure 9911. Calls between Site B IP phone and R2 PRI trunk should always use G711 codec. Score: 2 Points 2.3 CUCM SIP Trunks 2.3a-Inter-Site SIP Trunk using CUBE: HQ & Site B Configure R1 as Cisco Unified Border Element interconnecting HQ and SiteB CUCM clusters using SIP protocol. All 4 digit inter-site calls between HQ and Site B must go through R1, which must terminate media traffic, but pass through codec requested by endpoints. Make sure codec is use for all inter-site calls between HQ and Site B. For example when 2001 calls 300, the call must go through R1using ILBC codec Score: 3 Points 2.3b-CUCM SIP Trunk to PSTN Troubleshooting Build a SIP Trunk between the HQ CUCM cluster and the backbone CUCM (IP address 157.26.1.11) to enableH.264 video over IP between HQ Phone 2 and the PSTN phone. Create a single route pattern,85151111,on your CUCM and send the call over the SIP Trunk to the backbone CUCM. Use 8202xxxx, where xxxx is the 4-digit internal extension number, as the calling number for these calls. Do not send the calls through any intermediate routing identity (such as CUBE) Use RTMT in CUCM traces to find out why calls failed to complete through the SIP trunk to backbone CUCM. Find the following two SIP events and save the exact SIP messages in two separate notepad files on PC-1’s Desktop and name the files “SIP-TS-Event1” and “SIP-TS- Event2” correspondently. • Event 1: The SIP message where H.264 video capability is first passed. • Event 2: The SIP message where the call failed. Score: 4 Points 2.4 CUCM DIAL PLAN General Dial-plan Rules: Please note that due to the timed nature of the lab exam, this section is not designed to build a comprehensive and fully redundant dial plan between all sites. Instead, Candidates are expected to demonstrate their knowledge and experience by fulfilling a set of specific call-routing requirements. Call-routing configurations beyond these specific requirements will not be marked and will not result in additional points. Do not us the “@” wildcard in your CUCM dial plan. Country codes used in the lab are “1” (United States), “44” (United Kingdom), and “852” (Hong Kong). The PSTN access code at all sites is “9”. Country’s international access code is “011” for U.S and “00” for Hong Kong. Read the entire Call Routing section and understand all dialing requirement, before proceeding to configure your dial plans. 2.4a-HQ PSTN Dial plan The following call routing policies are in effect at HQ's PSTN Service Provider 1) HQ PSTN service provider mandates proper of both ISDN “called party number” and “Called party number type” (Subscriber, National and International) 2) Called party number types(local, National, International) must be set in ISDN setup "Called party number type" field for different types of calls (Local, National and international)*/ 3) Do not send leading digits in the ISDN called party number string, such as "1" for national and "011" for international to signal type of calls. Configure HQ CUCM to satisfy the following dial plan requirements. Access code for all PSTN call is 9 • Configure local route group for HQ's local Voice gateway : R1 • All HQ IP phones should be able to call local 7 digits PSTN numbers by dialing the access code "9"followed by 7 additional digits. The First digit after the access code could be any digit from 2to 9, the remaining digit could be 0 (zero) to 9. This type of call should always use the local gateway R1 as the primary route. 7-digit calling number (202xxxx where xxxx is internal extension) and calling name should be sent to PSTN. • All HQ IP phones should be able to place a national long distance PSTN calls by dialing the access code "9" followed by "1", followed by 3 digit area code and lastly 7 digit subscriber number. The First digit of the area code and subscriber number could be any digit from 2 to 9, the remaining area code and subscriber number digit could be 0 (zero) to 9. This type of call should always use the local gateway R1 as the primary route. 10-digit calling number (408202xxxx where xxxx is internal extension) and calling name should be sent to PSTN. Use 91206765XXXX where XXXX is any digit combinations, to test your HQ long distance calls, this call will ring pstn Line 1, the only exception to this rules is when the area code is 972. Please refer to question 2.4c for more information • All HQ IP phones should also be able to place international calls by dialing the access code "9" followed by "011", followed by variable length digit and ending with # sign. This type of call should always use local gateway R1. +1408202xxxx where xxxx is internal extension should be sent as calling number. Calling number should be sent to PSTN. All the above mention calls in the question must be serviced by single route pattern”\+!”. The exception to this rule are 911, Any H323 or SIP Trunks & VM Integrations numbers. Score: 4 points 2.4b-SiteB PSTN Dial plan The following call routing policies are in effect at Site B's PSTN Service Provider 1) SiteB PSTN service provider mandates proper of both ISDN “called party number” and “Called party number type” (Subscriber, National and International) 2) Called party number types (local, National, International) must be set in ISDN setup "Called party number type" field for different types of calls (Local, National and international)*/ 3) Do not send leading digits in the ISDN called party number string, such as "1" for national and "011" for international to signal type of calls. Configure Site B CUCM to satisfy the following dial plan requirements. Access code for all PSTN call is 9 • Configure local route group for Site B's local Voice gateway : R2 • All Site B IP phones should be able to call local 7 digits PSTN numbers by dialing the access code "9" followed by 7 additional digits. The First digit after the access code could be any digit from 2 to 9, the remaining digit could be 0 (zero) to 9. This type of call should always use the local gateway R2 as the primary route. 7-digit calling number (303xxxx where xxxx is internal extension) and calling name should be sent to PSTN. • All Site B IP phones should be able to place a national long distance PSTN calls by dialing the access code "9" followed by "1", followed by 3 digit area code and lastly 7 digit subscriber number. The First digit of the area code and subscriber number could be any digit from 2 to 9, the remaining area code and subscriber number digit could be 0 (zero) to 9. This type of call should always use the local gateway R2 as the primary route. 10-digit calling number (972303xxxx where xxxx is internal extension) and calling name should be sent to PSTN. • All Site B IP phones should also be able to place international calls by dialing the access code "9" followed by "011", followed by variable length digit and ending with # sign. This type of call should always use local gateway R2. +1972303xxxx where xxxx is internal extension should be sent as calling number. Calling number should be sent to PSTN. All the above mention calls in the question must be serviced by single route pattern”\+!”. The exception to this rule are 911 or any SIP trunk calls. Score: 3 points 2.4c-HQ Tail End Hop-OFF to SITEB Provision an H323 ICT trunk between HQ CUCM to SiteB CUCM to enable TEHO from HQ to Site B. When HQ Phones places a long distance call with an area code 972, the call should be sent over the IP Network to Site B CUCM which will ensure the call egress to local voice gateway. This call should connect using g729 codec. For example if HQ IP Phone 1 dials 919722522222 the call should first go through Site B t1 MGCP gateway as a local call, if this path R2 is used, calling number on the PSTN number should be 4082022001 along with appropriate calling name. If SiteB CUCM or R2 is unavailable to take the call, the call should be re-routed, use HQ's own gateway R1in this case, calling number seen on the PSTN phone should be 4082022001 and along with calling name. Score: 3 Points 2.4d-SiteB Plus Dialing Configure Site B CUCM to deliver the following globalized calling number to SiteB IP phones. 1) When HQ PSTN (PSTN Phone Line 1)calls Site B IP Phone by dialing 19723033001 and the call is not answered, the "missed calls" directory on Site B Phone 1 should display the missed call in a globalized format with “+” sign. " +14085151111". 2) When Site B PSTN calls Site B IP Phone by dialing 3033001and the call is not answered, the "missed calls" directory on Site B Phone 1 should display the missed call in a globalized format with “+” sign. " +19725252222". 3) When Site C PSTN calls Site B IP Phone by dialing 0019723033001 and the call is not answered, the "missed calls" directory on Site B Phone 1 should display the missed call in a globalized format with “+” sign. " +85225353333". 4) For any of the missed calls mentioned above, the owner of the Site B phone 1 should be able to press the "dial" soft key to place a call to appropriate destination through the Site B IOS MGCP Gateway R2 along with calling name. Score: 3 Points 2.4e HQ SIP Trunk redundancy to PSTN In Question 2.3b you build the SIP Trunk between the HQ CUCM cluster and the backbone CUCM (IP address 157.26.1.11) to enable H.264 video over IP between HQ Phone 2 and the PSTN phone. Since calls through this SIP trunk failed, the VoIP Service provider has enabled an alternate SIP trunk at 157.26.1.253 for you to try the same call. The route pattern is the same,85151111, however you must now originate your SIP request from R1 with IP address 142.102.64.254. Do not delete the first SIP Trunk from your HQ CUCM. Calls to 85151111 should always go to 157.26.1.11 first. Route the calls to alternate SIP trunk only when calls could not complete through 157.26.1.11. Continue to use 8202xxxx where xxxx is the 4-digit internal extension number and the calling number for these sites. Hints: - Turn on debugs on your devices to learn the SIP capabilities on the backbone side and adjust your configuration if necessary. You should be able to place a successful call with bi-directional H.264 video. Score: 3 Points 2.5-CUCM URI Dialing 2.5a-HQ Intra-site URI Dialing Enable HQ Phone 1 and HQ Phone 2 can call each other by dialing the end user’s URI. The URI for HQ Phone 1 [email protected] and URI id for HQ phone 2 is [email protected] Make the 5th button on each phone an URI speed dial button to the other phone and ensure the speed dial works between the two phones. Score: 3 Points 2.5b-HQ & SiteB Inter-site URI Dialing Configure ILS between HQ & SiteB CUCM clusters. HQ CUCM should be a ILS network hub while the Site B CUCM is a spoke. Make sure the clusters check for updates every minute. The URI for Site B Phone 1 is [email protected], while HQ URI’s were provided in question 2.5a. Add two speed dial entries on the 5th and 6th button of Site B Phone 1: the 5th button should call [email protected] and the 6th button should call [email protected] Add an additional 6th button on each HQ phone for Site B phone 1’s URI. Score: 3 Points 2.6-CUCM Media Resources 2.6a-HQ IOS Video Conference Bridge: R1 Configure and register R1 as Cisco IOS Video Conference Bridge for the HQ CUCM cluster. HQ Phone 2 should be able to invoke this conference bridge for a 3way video conference using H.264 code. After you properly register and provision routing for Site C Phone 2, make sure to verify that HQ Phone 2 can host a Video Conference with PSTN phone and Site C Phone 2. From HQ Phone 2, first place a call to PSTN Phone line 1 (85151111) and then conference in Site C Phone 2 at 4002. Score: 4 Points Section 3: Configure & Troubleshoot CISCO IOS UC Application & Features 3.1-CUCME IP PHONES 3.1a-CUCME SCCP IP PHONES Register and Configure SiteC Phone 1 according to the telephony number scheme. This phone should display globalized calling number, “+85224044001”, on the top right hand corner of the phone screen. Extension-to-Extension dialing at each site uses the last four digits only, calling name display should be delivered on internal 4-digits calls – use trivial caller names such as “SiteC Phone 1”. Make sure the phone displays the correct local time and that the date display is arranged as dd/mm/yy. Score: 2 Points 3.1b-CUCME SIP IP Phones Register and Configure SiteC Phone 2 according to the provided telephony number scheme. This phone should display globalized calling number, “+85224044002”, on the top right handside corner. Extension-to-Extension dialing at each site uses the last four digits only, calling name display should be delivered on internal 4-digits calls – use trivial caller names such as “Site C Phone 2”. Make sure the phone displays the correct local time and that the date display is arranged as dd/mm/yy. Lastly enable the video on this phone. Score: 2 Points 3.1c CUCME SIP IP Phone SecurityAdministration The SIP IP phone prompts administrator password on screen when anyone attends to navigate to the “Administration settings” page. The Default password is Cisco or CISCO. You are asked to remove this password and allow direct access to the administrator setting page. Score: 3 Points 3.2 CUCME Gateway Information below is relevant for your digital E1 gateway. • Line code/framing HDB3/CRC4 • ISDN switch type Primary- 5ess • Configure your gateways to take clock (layer1) from network. For PRI circuits configure your gateway as Layer 2 User side. • Refer to the individual gateway trunk-section for digit sending and receiving details. • Calling names should always be sent to the PSTN • Use full span on E1 PRI unless specified otherwise. • Gateway configuration must be verified with successful completion of inbound and outbound calls. Therefore, simple route pattern configuration is expected, even though call routing is not the key focus of this section. 3.2a- SiteC E1 – PRI Trunk Configure E1 PRI on the SiteC 2921 router R3, provision this router as H.323 gateway. Use only the first12 channels on this PRI circuit. This gateway should send and receive all H.323 traffic from its local voice VLAN interface with IP address of142.102.66.254. Telco delivers 8-digit Direct-Inward-Dial (DID) on inbound calls. Verify your configuration by placing an inbound call from PSTN phone line 4 to any DID numbers (2404xxxx) at Site C, where xxxx is any Site C internal 4-digit extension. Also make sure the outbound calls work by calling the local emergency number (999) for any Site C IP phones. For emergency calls, send 2404xxxx, where xxxx is 4-digit extension, as the calling number. Score: 2 Points 3.3- CUCME Dial plan 3.3a - SiteC PSTN Dial Plan Site C PSTN Service Provider routing policies. • Site C PSTN Service Provider requires proper setting of called party number string and called party number types (subscriber or international) for different types of calls (local or international) • Called Party number types (subscriber or international) must be set in ISDN set up messages. • For example, if Site C IP phones dials an US international number, 90014085151111, Site C’s PSTN would process the call only if “14085151111” was sent as the called number accompanied by called party number type of international. • Unknown called party number types is only accepted for emergency 999 calls. After you understand the Site C PSTN routing rules above, configure the following dial plan requirements for Site C. 1* All Site C IP phones should be able to place local PSTN calls by dialing the access code 9 followed by any 8 digit number string. These local calls should go out of local H.323 gateway R3. Send calling name display and 8-digit calling party number (2404xxxx) to PSTN. Calling party number type for these calls should be set to subscriber. 2* All Site C IP phones should be able to place international calls by dialing the access code 9, followed by 00, and then followed by country code and variable length digit patterns. For example, Site C users should dial 900442085554444 to reach the UK PSTN number. Some users dial # to signal end of dial and to avoid having to wait for the inter digit time out, while others do not. Send all international calls to R3. For international calls, send calling party number with Site C’s country code with (+8522404xxxx) and calling party number type as International, calling name should also be sent. Score: 2 points 3.3b-SiteC Inter-site calling with HQ & SiteB Configure R1 as Cisco Unified Border Element interconnecting HQ and Site B CUCM clusters with Site C using SIP protocol. All 4-digit inter-site calls between Site C and HQ or Site B must go through R1 using Sip, for example, when Site C Phone 2 calls Site B Phone 1 by dialing 2001 or vice versa, the call signal and media should traverse R1. All inter-site voice calls should use ILBC codec and make sure H.264 video works on calls between Site C Phone 2 and HQ Phone 2. Score:3 Points 3.3c-SiteC Mobile Connect:Single Number Reach Enable single number reach on SiteC Phone 1. Incoming call should ring both SiteC Phone 1 line 1 and the Site C PSTN number at 25353333. User should be able to answer the call from either Site C Phone1 or from PSTN phone. If the PSTN phone answers the call, Site C phone 1 should go into “Hold” mode. If the call is answered by the Site C phone 1, the PSTN phone should stop ringing. The user should be able to invoke mobile connect or send the call to mobile phone with the “Mobility” soft key. Score: 3 Points 3.4- CUCME Media Resource 3.4a- Hardware Conference Resources Register a Hardware Conference Bridge to the CUCME router, make sure the Hardware Conference resource can be invoked by either the Site C Phone 1 as well as Site C Phone 2. Verify a 3 party hardware conference, place a call from PSTN Phone. Score: 3 Points 3.5- Cisco Unity Express 3.5a- Site C CUE Initialization Your Cisco Unity Express Service engine has been restored to factory default. You will need to go through the one time service engine post installation configuration tool to initializing. Use the following information for Cisco Unity Express Initialization IP Address: 142.102.66.253 Hostname: CUE Domain name: ccie.cisco.com DNS: not necessary NTP:157.26.1.250 Time Zone: Hong Kong CUE web GUI account: administrator CUE web GUI account password: Score: 2 Points 3.5b - CUE and CUCME Integration Integrate CUE and CUCME using following information. • Voice mail pilot number 4220 • Set all voice mail account pins to 12345 Create account for Site C Phone 1 and Site C Phone 2. Make sure messages can be deposited and MWI works for local IP phones. Call should be forwarded to voice mail if the extension is busy or if an incoming call was not answered in 20 seconds. Make inbound PSTN calls can be forwarded to CUE as well. Score: 3 Points Section 4: Configure & Troubleshoot QOS and Security in Cisco Collaborations Solutions 4.1 QOS: Classification and Mapping 4.1a-Switch COS to DSCP Mapping Configure SW1 and Ethernet Switch module in R2 and R3 for the following COS to DSCP mapping for voice and interactive video traffic. • Voice signaling Cos3 to CS3 • Voice Media Cos5 to EF • Video Media Cos4 to AF41 Score: 3 Points 4.2 -QOS Congestion Management 4.2a Switch Traffic Marking and Policing Configure the following Voice and Video traffic management policy on voice ports with IP phone attached. • Allow a maximum of 3 G711 calls. Any excess voice media traffic should be dropped • Allow 24 kbps for voice signaling (SCCP and SIP. Any excessive signal traffic should be mark down to CS1. • Allow 5 MBPS for Video media, excess video media traffic should be mark down to CS1. • All data traffic should be re-marked to best effort Score: 4 Points SECTION 5:Configure and Troubleshoot Cisco Unity Connection 5.1- Cisco Unity Connection Integration 5.1a - CUC SIP Integration Integrate the HQ CUCM cluster and HQ CUC using SIP. Set the Voicemail pilot number to 2220. After the integration is completed you should be able to call into the voicemail pilot number from any HQ phone. Make sure the local PSTN phone can also place a call from line 1 to 2022220 and hear the CUC system greeting. Score: 3 Points 5.1b - CUCM SCCP Integration Integrate SiteB CUCM with HQ CUC using SCCP with the following information • Voicemail Pilot – 3220 • Voicemail ports – 3221-3224 • MWI On DN– 1998 • MWI Off DN– 1999 When Integration is completed, you should be able to call into the voicemail pilot number from Site B IP phone 1 and from the local PSTN line by dialing 3033220 Score: 2 Points 5.2 - Cisco Unity Connection administration 5.2a - Voicemail Provision for HQ users Create users for HQ Phone 1(hqone) and HQ Phone 2(hqtwo) on CUC. Set password to be “12345”. Also make sure the incoming calls to these phones, when not answered in 20secs or when there is an active call on the line, are forwarding to Cisco Unity Connection for voicemail service. Verify that you can leave and retrieve messages, and that MWI works. Score: 3 Points 5.2b-Voicemail Provision for SiteB users Integrate Site B CUCM with HQ CUC using SCCP with the following information Import the “sitebone” user from CUCM into CUC. Incoming calls for Site B phone 1 when not answered in 20 sec or when there is an active call on the line, should be forwarded to CUC. Verify you can leave and retrieve messages, and that MWI works. Score: 3 points Section 6 - Configure and Troubleshoot Cisco Unified Contact Center Express 6.1-Cisco Unified Contact Center Express integration 6.1a-HQ UCCX Integration Configure the following CUCM, UCCX and ICD script to accommodate the customer requirements for HQ listed below CTI Route Point – 2400 CTI Ports – 2401 to 2405 Jtapi Prefix - jtapi Jtapi password – cisco Rm username – rm Rm password – cisco Add a second line on HQ Phone 1 (DN 2201) and HQ Phone 2 (DN 2202) to be dedicated agent calls. Configure the IP phone agent phone service on both phones. Both agent should become ready for the next call immediately after the existing call is disconnected. Score: 4 Points Section 7 - Configure & Troubleshoot Cisco Unified IM & Presence 7.1 IM&P Integration 7.1a- HQ IM&P integration: CSF Soft Phone with Video Integrate HQ CUCM with IM and Presence services. Provision the jabber client on PC1 for HQ user “hqone”. Make sure the client can place audio phone calls with video using the high resolution video camera attached to the PC-1. Place a call to HQ Phone 2 and you should see the video streams between two endpoints. RDP into PC1 with the following credentials user name” admin” password” ” Refer to the following screen shot for provisional and sign on information Score: 4 Points 7.1b- HQ IM&P Integration: Desktop Phone Control Provision the jabber client on PC 2 for HQ user “hqtwo” which controls the desk 9971 HQ Phone 2. RDP into PC2 using the following credentials user name “admin” password “” Refer to the following screen shot for provisional and sign on information Score: 3 Points End of Question. Please if you have any of the following and willing to share, contact me so I can put it on the 1st post. I want as time goes on, this page should contain almost everything needed for CCIE Collaboration exams. 1. CCIE Collaboration Workbook by Collabcert 2, CCIE Collaboration Workbook by INE 3. CCIE Collaboration Workbook by IPExpert 4. CCIE Collaboration Workbook by Others 5. Complete CCIE voice Lab VMware ESXi images CUCM CUC CUPS UCCX 8.6 & 10.5 ( $65) 6. Complete Collaboration Lab walk-through video 7. Valid CCIE written exam 8. Lab questions 9. etc Please contact me if you have any thing you feel is useful. Cheers all. Rik.
  12. Hi everybody, Who one has cisco plm 11 version iso file? Please share it. Best regards.
  13. Hi Guys, I am putting Actualtest 400-051 written exam. Enjoy! [Hidden Content]
  14. INE - CCIE Collaboration Lab Preparation (Mark Snow) [Hidden Content] Hi. is it possible in mail.ru or cloud ? Thanks in advance
  15. hello iPexpert - CCIE Collaboration Written Exam VoD NextGen 21 videos - 1280x720 - flv - 3.64GB enjoy! [Hidden Content]
  16. Hi Memeber's Here are the dumps from CCIE COLLABORATION WRITTEN EXAM.Njoi!!![Hidden Content] 400-051.pdf
  17. Pls update ccie collaboration written from ipexpert
  18. Does anyone have this material???
  19. Anybody having latest work book form cciecollaborationlabs.com?
  20. Hi - Does any one in Hong Kong know where I can rent physical rack time in HK to practice for a few days before CCIE Collaboration attempt? Please advice if you know any rack rental provider, training center that can help. thanks
  21. Hi all, Need CCNP Collaboration training videos. CBT Nuggets or any other source. Thanks
  22. HI All, I'm about to re-certify my CCNP Collaboration. I've found my usual sites that I got exams / dumps no longer have Cisco exams. Where can I get reliable exams / dumps? I dont mind paying for them. But do you know what sites are reliable? Initially, I want to sit - 300-085 All help and advice appreciated. Thanks!
  23. Hi Guys, Can someone please help me understand what is the next step after i completed watching the Collaboration ATC videos of INE? I intend to use the INE's Rack rental but i am very confused about the study materials, i can't use the old CCIE voice workbooks since it has different topology. Thanks!!! Tom.
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