Voice CCIE labbing

Wednesday, April 14, 2010

Tuesday, April 13, 2010

Barge and cBarge on CUCM

To add Barge or cBarge :

Barge:

Device / Phone : turn Privacy to OFF (default is ON)
Device / Phone : set Built In Bridge to ON
Device / Phone : select Single Button Barge to either Barge or cBarge (default is OFF)
Device / Phone : select a Softkey Template. For Barge, a Standard User template is sufficient
Device / Phone : add a shared-line DN, i.e 2001 for both test phones

The 2 test phones will now have a second shared-line button. On test-phone1, use shared-line to dial to another number. Notice that test-phone2 second line will display a "double handset" sign on the shared-line and the shared-line button will turn RED. Press the RED button on test-phone2 to barge in

cBarge:

Device / Phone : turn Privacy to OFF (default is ON)

Device / Phone : set Built In Bridge to ON

Device / Phone : select Single Button Barge to either Barge or cBarge (default is OFF)

Device / Phone : select a Softkey Template. For cBarge, copy Standard User template to a new template and add cBarge softkey in call state "Remote In Use"

Device / Phone : add a shared-line DN, i.e 2001 for both test phones

Thursday, April 8, 2010

Tasks notes - important, go over

CUE and CME:

add voice translation-rule/profile for VM numbers
enable SIP - H323 interwork via "voice service voip"
bind SIP to interface gig0/0.230 (or same interface with CME)
enable SIP-UA
add Dial-Peer to make sure calls will reach CUE with SIP protocol and G711u - don't forget to include voice translation-profile
add Dial-Peer to make sure MWI numbers will use G711u
add num-exp (i.e num-exp 3500 21313500) to enable calls from PSTN to reach VM
add voicemail number in CME and call-forward in ephone-dn
add MWI on/off ephone-dn
add Transcoder into CME if VM access from other sites (HQ or BR1) is needed
use CUE Wizard for rest of task

Wednesday, April 7, 2010

AAR simplified in 5 minutes

In brief, AAR is for Automated Alternate Routing in case the WAN/IP network routing or bandwidth is restrictive to pass voice calls. Don't confuse this with SRST which is another subject

Procedure to setup AAR:

Decrease the bandwidth between sites so that a voice call wont have enough bandwidth, i.e change location bandwidth to 20K (minimum is 24K for g728 and 80K for g711).
on CUCM, Call-Routing / AAR-Group, add AAR-HQ, AAR-BR1 ...
Depend on the DialPlan configured previously, there 're 2 scenarios here for AAR routing.
If you configured global Route-Pattern \+! then adding Translation-Pattern to route calls , there 're no need to add Dial_prefix between sites since the global RP \+! should match all routes.
If you configured multiple RP for local/LD/INTL numbers, then calls using PSTN from HQ to BR1 should include prefix 9 (if BR1 External Phone Number Mask is 11 digits , i.e 1312301XXXX) or prefix 91 (if BR1 External Phone Number Mask is 10 digits , i.e 312301XXXX) and local calls inside HQ should use prefix 9 .
Update both HQ and BR1 Device/Phone AAR-CSS and AAR-Group with appropriate info.
Update both HQ and BR1 Device/Phone/Line AAR Settings with appropriate AAR Group.
Update both HQ and BR1 Gateway AAR-CSS and AAR-Group with appropriate info.

Test by changing Location bandwidth of BR1 site to 20K. Calls from HQ phone to BR1 phone should be using PSTN with message "Network Congestion - Rerouting" appears on the HQ phone.

Wednesday, March 24, 2010

IPIPGW from CUCM to BR2

Tasks: configure HQ gw as IPIPGW aka CUBE to handle SIP calls from CUCM to BR2 CME
Recall that CUCM is using a SIP-TRUNK and codec g711u while CME on the BR2 side is using H.323 and codec g729r8.

config HQ gw as IPIPGW with SIP interfacing CUCM and H323 toward BR2 CME

                                      voice service voip
                                         allow-connections h323 to h323
                                         allow-connections h323 to sip
                                         allow-connections sip to h323
                                         allow-connections sip to sip
                                      sip
   bind control source-interface Loopback0
                                           bind media source-interface Loopback0

   interface Loopback0
ip address 10.10.32.1 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip h323-id HQ-IPIPGW
h323-gateway voip bind srcaddr 10.10.32.1
   gateway
                                       sip-ua

config HQ gw with DSP resource to do transcoding. NOTE: IPIPGW binds to Loopback0 while DSP resources bind to Gig0/0.30. Test by calling from HQ toward BR2 PSTN and place both phones off-hook. If TRANSCODING is not working , will get busy tone when going off-hook.

                                    voice-card 1
   no dspfarm
      dsp services dspfarm
                                    sccp local GigabitEthernet0/0.30
                                    sccp ccm 10.10.30.1
                                    sccp ip precedence 3
                                    sccp
                                    sccp ccm group 1
   associate ccm 1 priority 1
   associate profile 1 register HQ-XCODER
                                    !
                                   dspfarm profile 1 transcode
maximum sessions 2
   associate application SCCP
                                    telephony-service
   sdspfarm units 1
   sdspfarm transcode sessions 2
   sdspfarm tag 1 HQ-XCODER
   max-ephones 2
   max-dn 4
   ip source-address 10.10.30.1 port 2000
   create cnf-files

config HW gw with a dial-peer to point H323 dialed string to BR2

                                  dial-peer voice 1000 voip
description ==== IPIPGW h323 to BR2 area code
destination-pattern 4423.T
session target ipv4:10.10.32.3
dtmf-relay h245-alphanumeric
   no vad
                                          codec g729r8
dial-peer voice 1001 voip
   description ==== abbrev. dialing to BR2 internal IP phones
   destination-pattern 3...$
   session target ipv4:10.10.32.3
   dtmf-relay h245-alphanumeric
no vad

Add a SIP-TRUNK on CUCM, this is SIP trunk from CUCM to HQ gw. Device-name = SIP-TRUNK-IPIPGW , DP = HQ , location = HQ , SIP destination-address = HQ Loopback0
Add this SIP-TRUNK into RG-BR2 .
Add RL-TEHO-HQ-TO-BR2 with 1st RG = RG-BR2 (Use Calling Party's External Phone Number Mask = ON , Discard Digits = predot) , 2nd RG = RG-HQ (Use Calling Party's External Phone Number Mask = ON , Discard Digits = predot , Prefix Digits (Outgoing Calls) = 011 ) .
Add Route-Pattern 9011.4423XXXXXXXX , partition = PT-HQ-INTL , Route-List = RL-TEHO-HQ-TO-BR2 , urgent priority , strip PREDOT .
For abbrev. dialing to BR2 internal IP phones , add following : RL = RL-TEHO-HQ-TO-BR2-INTERNAL with 1st RG = RG-BR2 (Use Calling Party's External Phone Number Mask = ON , Discard Digits = predot) , 2nd RG = RG-HQ (Use Calling Party's External Phone Number Mask = ON , Discard Digits = predot , Prefix Digits (Outgoing Calls) = 01144232131 ) . Pretty much same RL as on step 3 , except for the PSTN dialing part . Add Router-Pattern 3XXX , PT = PT-internal , RL = RL-TEHO-HQ-TO-BR2-INTERNAL

This should take care of HQ side, now we need to configure BR2 to handle incoming H323 calls

config BR2 as H323 GW

interface Loopback0
ip address 10.10.32.3 255.255.255.255
ip ospf network point-to-point
h323-gateway voip interface
h323-gateway voip h323-id BR2
h323-gateway voip bind srcaddr 10.10.32.3
gateway

config voice translation-rule & dial-peer to handle incoming calls from VOIP

                  voice translation-rule 1
   rule 2 /^44232131$3$$...$$/ /\1\2/    ==== strip to 3xxx for BR2 IP phones
   rule 3 /^4423$........$$/ /9\1/              ==== strip to 9xxxxxxxx for BR2 local calls
                  voice translation-profile teho-cucm-to-br2
translate called 1
                  dial-peer voice 2000 voip
description ==== incoming from CUCM
   translation-profile incoming teho-cucm-to-br2
   incoming called-number 4423.T
dtmf-relay h245-alphanumeric
   no vad
                        codec g729r8

Test calls by calling from HQ phones to BR2 internal phones + BR2 area PSTN numbers

config voice translation-rule and dial-peer in BR2 for abbrev. dialing to CUCM

                          voice translation-rule 2
                                rule 1 /1$...$$/ /7752011\1/
rule 2 /1$...$$/ /3123012\1/
                          voice translation-pro to-cucm
                                translate called 2
                          dial-peer voice 2001 voip
   description ==== from BR2 to CUCM
   translation-profile outgoing to-cucm
   preference 1
destination-pattern [1-2]...$
session target ipv4:10.10.32.1   ==== this is HQ Loop 0 int
dtmf-relay h245-alphanumeric
codec g729r8
no vad

config IPIPGW dial-peer to handle incoming calls from BR2

                        dial-peer voice 1001 voip
   description ==== BR2 calls to CUCM
      preference 1
   destination-pattern 775.T
   session protocol sipv2
   session target ipv4:10.1.200.21   ==== this is CUCM IP addrs
   dtmf-relay rtp-nte
   codec g711ulaw

config CUCM to handle incoming calls from BR2

                 add PT = PT-IPIPGW , CSS = CSS-IPIPGW (with PT-INTERNAL & PT-IPIPGW)
                 set SIP-TRUNK-IPIPGW with CSS-IPIPGW , significant-digits = ALL
                 add translation-pattern , pattern = 775201.XXXX , PT = PT-IPIPGW , CSS = CSS-IPIPGW

Additional tasks for BR2 to call HQ area code numbers

dial-peer voice 2002 voip ==== dial-peer on BR2
description ==== from BR2 to CUCM area code
preference 1
destination-pattern 9001775.......$
session target ipv4:10.10.32.1
dtmf-relay h245-alphanumeric
codec g729r8

no vad

voice translation-rule 1    ==== add voice translation-rule on HQ GW
        rule 3 /9001$.*$/ /\1/
voice translation-profile DID
translate called 1
dial-peer voice 1002 voip    ==== same dial-peer configured before on HQ
translation-profile incoming DID
incoming called-number .
on CUCM , add other PT such as PT-HQ-LOCAL , PT-HQ-LD into CSS-IPIPGW .
Add Translation-pattern , pattern = 775.XXXXXXX , PT = PT-IPIPGW , CSS = CSS-IPIPGW , discard-digit = PREDOT , Prefix Digits (Outgoing Calls) = 9 (remember , we want to use the HQ GW for PSTN local calls).

Test by calling from BR2 IP phone to HQ PSTN numbers + HQ IP phones.

CAVEATS: TEHO calls from BR2 to BR1 area configs:

on BR2, add voip dial-peer

                 dial-peer voice 3005 voip
                      translation-profile outgoing teho-to-cucm
                      destination-pattern 9001312.T
                      session target ipv4:10.10.32.1
                      codec g729r8
                      dtmf-relay h245-alphanumeric
                      no vad

on HQ gw, add voip dial-peer to forward incoming calls from BR2 to CUCM, but specify g729codec

                dial-peer voice 1004 voip
                      description ==== handle calls from BR2 , teho to BR1 area
                      destination-pattern 312.T
                      session protocol sipv2
                      session target ipv4:10.1.200.21
                      codec g729r8
                      dtmf-relay h245-alphanumeric

Test by calling from BR2 phones to BR1 area code number. Should see calls coming out of BR1 GW. Note: on CUCM, should set Region for G729 between HQ and BR1 (default is G711).

Tasks to go over before the LAB test - voice exam check list

DialPlan with GK - posted
DialPlan with IPIPGW - posted
DialPlan with COR - done
DialPlan with Local-route (single RP, translation-pattern, transformation-pattern)
CME with SCCP - done
CME with SIP endpoints - need to do SIP
CUE with CME - posted
CUE with CUCM - posted
Extension mobillity - posted
SNR - posted
SRST
AAR - posted
Call park, Call pickup, Barge, Callback - done Barge/cBarge, CallBack, callPark, callPickup = not enough phone
IPMA - posted
QOS -
IPCCEx scripting - need to do
https://learningnetwork.cisco.com/docs/DOC-7024

Tuesday, March 16, 2010

Integrating CUE and CUCM

CUE config:

interface Service-Engine2/0
ip unnumbered GigabitEthernet0/0.130
service-module ip address 10.10.130.3 255.255.255.0    >>> this is the IP addr of CUE
service-module ip default-gateway 10.10.130.1
ip route 10.10.130.3 255.255.255.255 Service-Engine2/0
>>> no other config on CUE or IOS router is necessary

Log into CUE module and restore to factory default
BR1#service-module in2/0 sess
    ....
NME-CUE#
NME-CUE# offline
    ....
Are you sure you want to go offline[n]? : y
    ....
NME-CUE(offline)# restore factory default

NME-CUE(offline)#
MONITOR SHUTDOWN...
                    After a few minutes, CUE will boot up, fill in the necessary info
Do you wish to start configuration now (y,n)? y

       .................

Enter Call Agent
(CME, CCM, or enter to use CME as default): CCM                         >>>> make sure you select CCM
Selected Call Agent: CCM
Setting Call Agent to CCM in /usr/wfavvid/workflow.properties
    ....
    ....
Enter administrator user ID:
(user ID): admin
Enter password for admin:
(password):
Confirm password for admin by reentering it:
(password):

SYSTEM ONLINE
br1-cue# sh software license
    ....
Core:
- Application mode: CCM
- Total usable system ports: 8    >>>CUE has 8 lic ports, make sure on CUCM, you will add 8 CTI ports ..... hmmmm, maybe 2 CTI ports is good enough . Verified operation on 3/29 with 2 CTI ports.
    ....
br1-cue#

Add CTI port on CUCM, under Device/Phone, add new Phone Type = CTI Port, device name = cue1, DP = BR1, location = BR1. Add new DN for the CTI Port, DN = 1601, PT = PT-USA . Repeat for all ports of CUE. Verified operation on 3/29 with 2 CTI ports.

IMPORTANT : don't forget to config "External Phone Number Mask" for CUE CTI ports and Route-Point , or they won't register in CUCM

Add CTI Route Point on CUCM, under Device/CTI Route Point, add new CTI RP, device name = BR1-CUE , DP = BR1, location = BR1 , CSS = CSS-GLOBAL. Add new DN for the CTI RP , DN = 1600, PT = PS-USA, CSS = CSS-GLOBAL.

Add new application user, username = cue , password = cisco , Controlled Devices = cue1, cue1 .... cue8 , BR1-CUE , Groups/Roles = standard CTI enabled, CTI control of all devices, AXL API access. This username = CUE JTAPI username. Controlled Devices = BR1-CUE + cue1 + cue2 ...

Create new Voicemail Pilot and Profile for CUE - on CUCM , create CUE Voicemail Pilot (DN=1600) and CUE Voicemail Profile . Assign this Voicemail Profile into IP phone (i.e. BR1 Phone1 , br1user1 ...) which wants to use CUE instead of Unity.

- Web login into CUE and run the Wizard , Web User Name:= admin/cisco , JTAPI User Name = cue/cisco. Alternately, you can use cue/cisco & cue/cisco for both Web and JTAPI usernames. Import CUCM users and complete Wizard setup.

- Reload CUE. Re-login into CUE and verify imported users has "Primary Extension , i.e 2001, 1003

- Test by dialing into the IP phone (BR1 phone1) which was setup with the CUE Voicemail Profile above, should get the CUE prompt and MWI led should lit up. If a fast BUSY tone is heard when dialing CUE pilot # , there 're a mismatch of codec between sites (remember that CUE uses SIP and G711 only) . Either change codec relationship in System/Region or add in required Transcoder.

Screen shot of CTI Ports and CTI RP on a working CUCM. Notice that both CTI Ports & CTI RP are registered to CUCM with CUE IP addrs (10.10.130.3)