(2) New Mgcp Lab

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CCIE VOICE LAB TOPOLOGY

HQ DN 2001 2002 2003 2122 2121 2887 SB DN 3001 3002 3124 SC DN

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4001 4002

NOTE: All the users are by default created in the lab

IE J V U O N IC E E 20 LA 0 B 9 S .CModel Username Password cisco cisco cisco Pin 7960 manger 7960 assistant 7960 jayiay FXS on main site Router 1/0/1 Vg248 port 1 phone Vg248 port 3 fax 12345 12345 12345 Model 7960 7960 ATA Username Password Pin Model 7960 7960 Username Password Pin

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1. Basic Campus Design Configure VLAN on SW1, SW2, R2 for HQ, SB, SC according to tables. Provide Windows DHCP for HQ & SB for voice and use CCMP as TFTP server. Range is from 142.1XX.64.10 to 142.1XX.64.30. Use SC Router as IOS DHCP for SC. Range is from 142.1XX.66.10 to 142.1XX.66.30. Synchronize CCMP, CCMS and routers with NTP server on address 156.26.1.70. Do not set NTP on Unity.

SB Voice Vlan Data Vlan Loopback

HQ Voice Vlan Data Vlan Loopback SC Voice Vlan Data Vlan Loopback

PSTN Phone Number

IE J V U O N IC E E 20 LA 0 B 9 S .CVlan Number 3XX XX3 Vlan Name SiteB_Voice3XX SiteB_DataXX3 DN 9723YY3XXX 9723YY3XXX Vlan Number 1XX XX1 Vlan Number 4XX XX4 Vlan Name HQ_Voice1XX HQ_DataXX1 Vlan Name SiteC_Voice4XX SiteC_DataXX4 DN 4082YY2XXX 4082YY2XXX DN 85224YY2XXX 85224YY2XXX Line 1 Line 2 Line 3 Line 4 Line 5 40851588YY HQ PSTN 40851588YY 97252588YY SB PSTN 97252588YY 852253588YY SC PSTN 852253588YY 44 UK PSTN 44

Interface IP 142.1XX.65.254 142.2XX.65.254 142.XX1.65.254 Interface IP 142.1XX.64.254 142.2XX.64.254 142.XX1.64.254 Interface IP 142.1XX.66.254 142.2XX.66.254 142.XX1.66.254

Description

190055588YY PSTN 190055588YY

NOTE: All the vlans and ip address are already create just need to verify the same.

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911 Emergency 911

CME AND CALL MANAGER BASICS

2. Call Manager and Call Manager Express Configure HQ and SB phones, according to tables. Configure VG248 and ATA on HQ, according to tables. Port 2 of the ATA must register as per the given snap. Configure SC phones, according to the tables. Phones should looks like it is shown in the pictures. Configure phone 1 with DN of 4001 for primary line and second line with DN 4101. When someone calls 4001 when it is in call, it must provide call waiting option for the second call. For 4101 when someone calls and it is in call, must provide fast busy tone. Configure phone 1 with DN of 4002 for primary line and second line with DN 4102. For both lines when someone calls the line and it is in call, must provide fast busy tone. Configure speed dial as per the given snap of IP phone on line 3 Disable pc port for all cme phones and ccm phone 3 Phones should look like shown in the pictures.

Extend forward no answer in ccm to 20 secs using systemwide parameters.

Also, make sure that if there is active call on any phone, second caller should hear ring. Persist same behavior when you toggle between hold and resume.

hh:mm mm/dd/yy

Your current options Redial Newcall hh:mm mm/dd/yy

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Your current options Redial Newcall

IE J V U O N IC E E 20 LA 0 B 9 S .C24XX4001 4001 Sales Line 1 xtransfer to VM 4001 DND CfwdAll 24XX4002 4002 Sales Line 2 DND

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GATEWAYS 3. Voice Gateway and signaling All interfaces are ISDN PRI. All gateways must use channels from 1 to 6. T1 switch type is Primary-NI2; framing ESF; linecoding B8ZS. E1 switch type is Primary-Net5; framing is CRC4; linecoding HDB3.

You can test that T1 is working, by dialing 911 for emergency services.

You can test that T1 is working, by dialing 911 for emergency services.

Configure SC to use E1 on router configured as H323 gateway with CME for connection to PSTN. 8 digits caller ID and caller name should be sent to the PSTN.

NOTE: First find out about glare. "Means if Provider will be sending the call from top channels" we should keep bottom-up

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You need to capture channel status of PRI on HQ and BR1 gateway and save the file on subscribers desktop. Glare Use Perfmon to eliminate chances of glare.

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Configure SB to use T1 on router configured as MGCP gateway for connection to PSTN. Caller ID and caller name should be sent to the PSTN.

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Configure HQ to use port 3/1 on 6608 for T1 connection to PSTN. Caller ID and caller name should be sent to the PSTN.

CALL ROUTING 4. Call Routing All users can dial any number, except other is not specified later in the section. Except other is not specified all calls to PSTN in Site A and B must present 10 digits calling number to PSTN and 8 digits in Site C and also must present names. All calls to PSTN must be dialed using prefix 9. Names must be displayed on incoming calls from PSTN. Use descriptive names (for example Site B Phone1). HQ Calls

Local calls from HQ are dialed as 7 digit number starting with digits from 2 to 9. PSTN expects 7 digits. Use 6608 gateway as first priority and SB as second priority. Long distance calls from HQ are dialed as 11 digit number starting with digit 1, second and fifth digits must be from 2 to 9. PSTN expects 11 digits. Use 6608 gateway. No redundancy is required. Route calls dialed from HQ users as local or long distance number belonging to DID range of HQ, as an internal call. Called users must see 4 digits internal call number. Make route pattern *1239 only for local local for both the sides (HQ and CME) so so that when user will use this route pattern it should not wait them 15 sec

Solution for *1239 is we need to click on urgent priority for this route pattern SC Calls

Local calls from SC are dialed as 8 digit number starting with digits from 2 to 8. PSTN expects 8 digits. Route local calls from Site C through local gateway only. PSTN expects 8 digits as calling number. Route international calls from Site C through local gateway only. International calls are dialed with prefix 00 followed by variable length number. PSTN expects 8 digits as calling number. Emergency is 999 , Local call is 8 digit 9[2-9]XXXXXXX, LD is 11 digit and international is 001 Calls to CME to CCM will be 4 digit dialing and calling number will be same, If GK fails call should be automatically routed to E1 on CME confgiure a call block rule to block international numbers (900T), allow phone 2 to override the block by dailing 12345. After 7PM the phones logged in should automaticallay get logout out.

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GATEKEEPER 5. Gatekeeper Gatekeeper configuration should look this! They will provide 3 snaps in the lab. Only 4001 and 4002 should get register with GK 1) Configure the router at the Main site as the gatekeeper. Gatekeeper should be configured so that the output from the command show gatekeeper endpoints looks like is shown in the picture: SHOW GATEKEEPER ENDPOINT SNAP

Voice Capacity Max.= Avail.= 0 Current.= 3 Total number of active registrations o o The domain name is ccievoice.com. The tech-prefix for HQ-Trunk and CME-HKG is 1#.

You are not allowed to use default tech prefix, zone subnet and static alias commands! No e-164 numbers should register in gatekeeper.

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GATEKEEPER ENDPOINT REGISTRATION ======================================================================== CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags ------------------------------------------------------------------------------------------------------------------------------142.XX.64.11 1720 142.XX.64.11 551 16 GK VOIP-GW H323-ID: HQ Trunk_1 142.XX.64.12 1720 142.XX.64.12 551 18 GK VOIP-GW H323-ID: HQTrunk_2 142.XX1.66.254 1720 142.1XX.66.254 634 02 GK H323-GW H323-ID: CME-HKG 4001 4002

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2) Gatekeeper should be configured so that the output from the command show gatekeeper gw-type-prefix looks like is shown in the picture:

SHOW GATEKEEPER GW-TYPE-PREFIX SNAP

Main site phones should be able to call to site C phones using 4 digits dialing. Calls should be routed first through HQ-GK, then local gateway as backup. Called users must see 4 digits as a calling number. Site C phones should be able to call to HQ phones using 4 digits dialing. Calls should be routed first through HQ-GK, then local gateway as backup. Called users must see 4 digits as a calling number. No need to making call from site B to site C, or Site c to Site B using 4 digits

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GATEWAY TYPE PREFIX TABLE ========================= Prefix: 1#* Zone GK master gateway list: 142.XX.64.11:1720 HQ-Trunk_1 142.XX.64.12:1720 HQ-Trunk_2 142.1XX.66.1:1720 CME-HKG Zone GK prefix 4* priority gateway list(s): Priority 10: 142.XX1.66.1:1720 CME-HKG Zone GK prefix 2* priority gateway list(s): Priority 10: 142.XX.64.12:1720 HQ-Trunk_2 Priority 9: 142.XX.64.11:1720 HQ-Trunk_1

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GK Advanced NOTE: There is 2 different scenarios coming in the lab you will get scenario 1 or scenario 2 1) While dialing from HQ to CME if any call get reject due to any reason it should reroute to HQ PSTN. 2) While dialing from HQ to CME only 1 call should handle Subscriber and another call should automatically reroute to HQ PSTN number.

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6. Codec Selection and CAC The codec for the calls within the same site should be used G711, and calls to others sites should be G729. Only 4 calls are allowed the site B and the Main Site. Use only g.729 codec for calls to the gatekeeper.

7. Media Resources Management NOTE:

1) There was no transcoder in HQ site. 2) NO MOH/ multicast asked. 3) CME transcoder it was asked to use all unused dsp resources for transcoding. 4) Capture the screen from perfmon and save it on desktop as cfb.msc

Achieve load balancing between software conference bridges and use perfmon to generate the output as follows ,

For CME Transcoder question:-

Use unallocated dsp resources on cme router for transcoding.

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8. QoS Features

There is a 384kbps link between Main Site and Site B and 768kbps link between Main Site and Site C. Configure LFI MLPPP between Main Site and Site B. Do not use MLP between HQ and SC. LLQ o

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Voice Media: Calculate 6xG729 calls between sites. Other Traffic: user fair queue.

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Configure RTP compression for voice media between Main Site and Site B only. You cannot configure it directly on the interface. Every other traffic should go to class default and should be policed so all traffic that is equal or less than 60 % from the PVC must be transmitted and all other traffic must be remarked to 0.

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Voice Signaling: Calculate 10% from PVC for signaling, but you cannot use bandwidth percent command to achieve this.

9. Voice Mail Integration Unity Basic. o o Unity username and password are Username: administrator and Password: ccievoice Configure Main Site Phone 1 (2001), Phone 3 (2003) and Site B Phone 1 (3001) have voicemail box. Unity domain: ccievoice.com VM Pilot: 2220 VM Ports: 2221 - 2223 MWI ON/OFF: 1998 / 1999

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Unity

1) Unity page will not open in the lab you need to grant access for getting the page.

Solution :- You need to go in commserver directory in command prompt and enter the command Grantuseraccess.exe u CCIEVOICE\Administrator s EAdmin

3) Grant VM access to all the phones in HQ and SiteB phones

Unity Advanced

VPIM

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IE J V U O N IC E E 20 LA 0 B 9 S .CAll Phone Passwords must be 12345 o o o o o o o CUE Doman name: cisco.com Unity Doman name: ccievoice.com DNS - 142.xx.64.13 NTP 156.26.1.70 Unity ID 11X CUE ID 852

All necessary installations are done. You must enable VPIM and configure it from HQ to SC and the opposite site.

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Configure VPIM between Unity and CUE = Unity id is 1YY and CUE is 852 IMP Note :- You should know how to configure VPIM from ip as well as domain name bec they can ask anything its depend on the POD to POD

VPIM BROADCASTING When user send the message from Unity to cue all the cue phone should get the broadcast message as well as Hq phone 1 and phone 2 will get the same broadcast message make sure the MWI is lit CUE Advanced

Configure auto attendant so that if someone call on 4010 it should say Welcome to CCIE Voice lab and if someone press 0 it should got to phone 1 voicemailbox.

CUE integration with CME. Configure mailboxes for both ip phones at siteC. Record standard greeting for both the users.

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10. CRS Voice Applications ICD Basic o Configure ICD for HQ Phone 3. Create second line on the phone with DN of 2103, and use this phone as agent phone. Agent Username: jayjay Agent Password: cisco JTAPI username: jtapi Password: cisco Rm username: rm Password: cisco

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ICD Advanced o

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IE J V U O N IC E E 20 LA 0 B 9 S .CPilot Number: 2400 Ports: 2501 2503 IPCC url they will provide they wanted users should use single button login

Change the ICD script, Requirement is only 1 call should be in queue at a time rest all the calls should directly transfer to VM of agent (2103), When someone will the message 2103 should get only envolvope and MWI should not be lit.

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Solution

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11. CCM Applications Only Make sure MWI will work for manager phone

Manager Phone 2001 2111 Intercom Assistant #

Assistant Phone

IPMA console must be installed on CCM Subscriber

NOTE: Line 2100 should be shared between assistant and Manager line and it should use software

IE J V U O N IC E E 20 LA 0 B 9 S .C2100 2002 2102 2112 Intercom Manager # 2100

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12. NEW Directory service and speed - dial 1) Users from co-operate directory should not view CRSAdmin user in the ip phone 2) When CME user press the local speed dial they should view 2 numbers (they will provide a snap)

Solution for 1st point

http://cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00804d2087.shtml

Note: After you run the script, the user is hidden in the corporate directory and the CCMadmin user page. See the Unhide User section of this document in order to add the user back. Complete these steps in order to hide users in DC Directory:

1. Open Notepad on the publisher server. 2. Cut and paste these four lines and save the file as a text file on the C: drive as hideuser.ldif. Note: The spaces must appear as this output shows. Ensure that there are no extra spaces in the LDIF file. Otherwise, the script fails when you run it. dn: cn=[userid], ou=users, o=cisco.com changeType:modify replace:Description Description:CiscoPrivateUser

3. Set the [userid] to be the user you want to hide. For example: 4. dn: cn=ctiuser, ou=users, o=cisco.com 5. changeType:modify 6. replace:Description Description:CiscoPrivateUser

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7. Choose Start > Run > cmd in order to open a command prompt on the publisher server, and run this script in order to set the description field in DC Directory: 8. ldapmodify -h -p 8404 -D "cn=Directory Manager,o=cisco.com" -w -c -f hideuser.ldif.txt This output shows a successful run: C:\>ldapmodify -h CallManagerPub -p 8404 -D "cn=Directory Manager, o=cisco.com" -w Cisco -c -f hideuser.ldif.txt modifying entry cn=ctiuser, ou=users, o=cisco.com

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Hide Users in DC Directory

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Solution for 2nd point http://cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmespeed.html Enabling a Local Speed Dial Menu: Example The following commands enable the Cisco web browser and set the HTTP path to flash memory so that the speeddial.xml file in flash memory is accessible to IP phones: ip http server ip http path flash:

Local Speed Dial Record 1 to 1 of 1 Security 71111 Marketing 71234 Tech Support 71432

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The following XML filespeeddial.xml, defines three speed-dial numbers that will appear to the user after they press the Directories button on an IP phone.

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13. SRST

Allow only phone 1 and phone 3 of BR1 should get register with SRST New Call softkey should be available on ATA during hold. 3002 is not on SRST. If someone calls 3002, call should be transferred to 3001 after 12s no answer, call should be forward to voicemail go to voicemail of 3002, caller should hear 3002 is not available..3001 should ring. You cannot use translation and num-exp to achieve this.

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