TO-DO

test123

Barge and cBarge on CUCM

To add Barge or cBarge :

1. 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

1. 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

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 cbarge in. Note that cBarge use CFB resource , therefore MRG / MRGL must be set up and assigned to Device/Phone.

Tasks notes - important, go over

1. 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

1.  

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:

1. 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).
2. on CUCM, Call-Routing / AAR-Group, add AAR-HQ, AAR-BR1 ...
3. Depend on the DialPlan configured previously, there 're 2 scenarios here for AAR routing. 
4. 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.
5. 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 .
6. Update both HQ and BR1 Device/Phone AAR-CSS and AAR-Group with appropriate info. 
7. Update both HQ and BR1 Device/Phone/Line  AAR Settings with appropriate AAR Group.
8. 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.

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.

1. 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

1. 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

1. 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

1. 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
2. Add this SIP-TRUNK into RG-BR2 . 
3. 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 ) .  
4. Add Route-Pattern    9011.4423XXXXXXXX  , partition = PT-HQ-INTL , Route-List = RL-TEHO-HQ-TO-BR2 , urgent priority , strip PREDOT .
5. 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

1. 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

1. 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

1. 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

1. 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

1.   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

1. 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  

1. 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 .
2. on CUCM , add other PT such as PT-HQ-LOCAL , PT-HQ-LD into CSS-IPIPGW . 
3. 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

1. DialPlan with GK - posted
2. DialPlan with IPIPGW - posted
3. DialPlan with COR - done
4. DialPlan with Local-route (single RP, translation-pattern, transformation-pattern)
5. CME with SCCP - done
6. CME with SIP endpoints - need to do SIP
7. CUE with CME - posted
8. CUE with CUCM - posted
9. Extension mobillity - posted
10. SNR - posted
11. SRST
12. AAR - posted
13. Call park, Call pickup, Barge, Callback - done Barge/cBarge, CallBack, callPark, callPickup = not enough phone
14. IPMA - posted
15. QOS -
16. IPCCEx scripting - need to do
17.  
18.  
19.  
20. https://learningnetwork.cisco.com/docs/DOC-7024

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#

1. 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

1. 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.

1. 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 ...

1. 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)

CUCM dialplan with GK installed on the HQ router

Tasks to accomplished:
   1. Calls from HQ to BR2 should be made over GK trunk with PSTN as backup
   2. Calls from HQ to BR2 area/city codes  should be made over GK trunk with PSTN as backup
   3. Calls from BR2 to HQ should use 4 digits dialing via GK trunk with PSTN as backup
   4. Calls from BR2 to HQ area/city codes should be made over GK trunk with PSTN as backup 

HQ number = 17752011001
BR2 number = 442321313001 (w/ 011 as international code)

• config BR2 as 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 id ZONE-01 ipaddr 10.10.32.1 1719  ==== ip addr of HQ Loop0
 h323-gateway voip h323-id BR2
 h323-gateway voip tech-prefix 1#
 h323-gateway voip bind srcaddr 10.10.32.3
gateway      ====  don't forget to turn on gateway feature

• config HQ for GK

gatekeeper
 zone local ZONE-01 cisco.com 10.10.32.1  ==== define GK id
 zone prefix ZONE-01 1* gw-priority 9 TRUNK-GK-HQ_1
 zone prefix ZONE-01 1* gw-priority 0 BR2   ==== avoid sending calls with prefix 1 to BR2
 zone prefix ZONE-01 44* gw-priority 9 BR2
 zone prefix ZONE-01 44* gw-priority 0 TRUNK-GK-HQ_1  ==== avoid sending calls with prefix 44 to CUCM
  zone prefix ZONE-01 1... gw-priority 10 TRUNK-GK-HQ_1  === this is for HQ internal phones
  zone prefix ZONE-01 2... gw-priority 10 TRUNK-GK-HQ_1  === this is for BR1 internal phones
 no shutdown

Configure following for those 4 tasks:

1.    Task #1 -  Calls from HQ to BR2 should be made over GK trunk with PSTN as backup

• In CUCM, add GK with HQ interface Loop0 IP address

• add h.225 (GK controlled) trunk

              - device name = TRUNK-GK-HQ
              - device pool = HQ
              - Inbound Calls , Significant digits = 4   (so that calls from BR2 to HQ will be stripped to 4 digits)

              - Gatekeeper Name = HQ interface Loop0 IP address
              - Terminal Type = gateway
              - Technology Prefix = 1#
              - Zone = ZONE-01

• add GK trunk into a RG
• add RG into RL , name = RL-GK-TO-BR2,  first RG = RG-BR2 , Prefix Digits (Outgoing Calls) = 1#44232131 ,   secondary RG = RG-HQ with Prefix Digits (Outgoing Calls) = +01144232131   RG-HQ served as PSTN backup from HQ to BR2
• add Route Pattern, Route Pattern = 3XXX, Partition = PT-INTERNAL or PT-USA , Gateway/Route List = RL-GK-TO-BR2 .

That should handle outgoing calls from CUCM HQ to BR2. Next, we need to configure BR2 router to handle incoming calls from GK
    voice translation-rule 200
      rule 1 /1#44232131\(....\)/ /\1/
      rule 2 /1#4423\(........\)/ /9\1/
      rule 3 /1#0114423\(........\)/ /9\1/
    voice translation-profile GK-INCOMING
      translate called 200
   dial-peer voice 2000 voip
      translation-profile incoming GK-INCOMING
      incoming called-number 1#.T
     dtmf-relay h245-alphanumeric
      no vad
      codec g729r8

2.   Task #2 -  Calls from HQ to BR2 area/city codes  should be made over GK trunk with PSTN as backup

• Add new RL , name = RL-TEHO-HQ-TO-BR2
• First RG = RG-BR2,  Discard Digit = predot , Prefix Digits (Outgoing Calls) = 1# (remember that user will dial 9 011 4432 XXXXXXXX  for BR2 city/area code)
• Second RG = RG-HQ , Discard Digit = predot trailing # 
• Add new Route Pattern, pattern = 9.0114423XXXXXXXX# , Gateway/Route List = RL-TEHO-HQ-TO-BR2

3.   Task #3 and #4 - Calls from BR2 to HQ/BR1 should use GK trunk with PSTN as backup. Notice that BR2 user will dial 9001775....... for HQ area code numbers and 1XXX for HQ internal phones.

• on CUCM , add PT , CSS , to handle incoming call from BR2 . PT = PT-GK-TRUNK , CSS = CSS-GK-TRUNK with PT-INTERNAL + PT-GK-TRUNK + PT-HQ-LOCAL + PT-HQ-LD _ PT-BR1-LOCAL + PT-BR1-LD
•  Add Translation-Pattern  1#1775.XXXXXXX  (pt = pt-gk-trunk , css=css-gk-trunk). 
  Discard PREDOT , prefix = 9 . Repeat for BR1 area code
• Add Translation-Pattern  1#.XXXX  (pt = pt-gk-trunk , css=css-internal) . Discard PREDOT.  This is for HQ and BR1 internal phones
• Change TRUNK-GK-HQ with CSS = CSS-GK-TRUNK , Significant Digits = ALL

• configure BR2 dial-peer for abbrev. dialing to 1XXX & TEHO dialing to HQ area

                                        voice translation-rule 2    ==== strip out 900
                                                  rule 1 /9001775\(.*\)$/ /1775\1/
                                                  rule 2 /9001312\(.*\)$/ /1312\1/
                                        voice translation-profile teho-to-hq-area
                                                 translate called 2
                                       dial-peer voice 1004 voip
                                                description ==== teho to HQ area code
                                                translation-profile outgoing teho-to-hq-area
                                               destination-pattern 9001775.T
                                                session target ras
                                                tech-prefix 1#

                                     dial-peer voice 2002 voip
                                                description ==== abbrev dialing to HQ
                                                destination-pattern 1...$
                                                session target ras
                                                tech-prefix 1#
                                               dtmf-relay h245-alphanumeric
                                               no vad
                                    ! PSTN backup dial-peer
                                    dial-peer voice 3001 pots
                                              description ==== abbrev. dialing to HQ
                                              prefer 2
                                             destination-pattern 1...$
                                              port 1/0:15
                                             prefix 9001775201

CME MWI interwork with CUE

Remember that CME signaling is SCCP in nature and using G729r8 while CUE only supports SIP and G711u.
To get MWI working between CME and CUE, perform following:

! dont forget the ever-important voice translation-rule

voice translation-rule 400
 rule 1 /^.*\(3500\)$/ /\1/
 rule 2 /^.*\(3555\)$/ /\1/
voice translation-profile TO-VM
 translate calling 400
 translate called 400
 translate redirect-target 400
 translate redirect-called 400
!

! enable SIP and H323 intersignaling

 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 GigabitEthernet0/0.230    ====  do not use Loopback0

  bind media source-interface GigabitEthernet0/0.230      ====  or MWI won't work

! add a dial-peer to make sure calls to VM is using SIP and G711u

dial-peer voice 3500 voip
 description ==== to VM
 destination-pattern 3500
 session protocol sipv2
 session target ipv4:10.10.230.3
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

! add dial-peer to handle calls from PSTN to VM

dial-peer voice 21313500 voip
 description ==== VM
 translation-profile incoming TO-VM
 translation-profile outgoing TO-VM
 max-conn 3
 destination-pattern 21313500
 session protocol sipv2
 session target ipv4:10.10.230.3
 dtmf-relay sip-notify
 codec g711ulaw
 no vad

!  add this dial-peer to make sure MWI calls from CUE to CME is using G711u, otherwise MWI wont work

dial-peer voice 3998 voip
 description ==== MWI on and off
 incoming called-number 399[8,9]....
 codec g711ulaw
 no vad
!
num-exp 3500 21313500
num-exp 3555 21313555
sip-ua   
 disable-early-media 180
! the usual CME config
telephony-service
 max-ephones 10
 max-dn 40
 ip source-address 10.10.230.1 port 2000
 dialplan-pattern 1 21313... extension-length 4 no-reg
 voicemail 3500

!  add MWI number , notice that MWI is in this format XXXX.... (XXXX = mwi number, .... = user number)

ephone-dn  5
 number 3998....
 mwi on
!
ephone-dn  6
 number 3999....
 mwi off

Extension Mobility made easy - 20 minutes

Cisco Extension Mobility allows users to temporarily access their Cisco IP Phone configuration such as line appearances, services, and speed dials from other Cisco IP Phones.

1. Add new phone service , name it "Extention Mobility" - Device/Device Setting/Phone Services
 Use following URL for ext-mobility : http://yourCUCM_ip_addr:8080/emapp/EMAppServlet?device=#DEVICENAME#  . You can get this URL from the CUCM help menu

2. Add new "Device Profile" for your IP phone , let 's name it "EM-7970" under menu Device/Device Setting/Device Profile . This device profile is dependent on the type of IP phone you have , ie. 7970 , 7960 ...  So if you have 2 different types of IP phones, you need to create 2 separate Device profiles, i.e EM-7970, EM-7960 ... Add DN to this Device-Profile, for ex: DN=1002, PT = PT-internal. Also need to subscribe this "Device Profile" to Ex-Mo service (otherwise user cannot log out of Ex-Mo).

3. Configure end-user - under menu User Management/End Users, add new user or open an existing user, select Extension Mobility Available Profile "EM-7970" (added in step 2). Make sure the usual end-user groups (CTI enabled, CCM user ...) are configured, "allow control of device from CTI" BOX is checked and user is associated to a Phone (Controlled Device).

4. Configure IP phone - under menu Device / Phone, select a phone.  Checked on BOX labeled "Enabled Extention Mobility" and select Log out profile to "EM-7970" (configured in step 2).  Subscribe phone to "Extention Mobility" service (configured in step 1).  Reset phone as needed.

5. To test EM, on a same type IP phone , i.e. 7970, select service button. Phone should prompt with UserID and PIN # to log in. Once user is successfully logged in, phone will reset with correct softkey and button template.

ISDN L2 on HQ did not came up after configure MGCP

On this particular instance whereas after I configured ISDN MGCP on the HQ router and ISDN layer2 did not came up. After checking with the obvious CLI such as "show mgcp endpoint , show mgcp status , show ccm-manager host"  and realize everything looked good. Decided to reload the router and that fixed the problem
              
HQ#sh isdn sta
Global ISDN Switchtype = primary-ni
%Q.931 is backhauled to CCM MANAGER 0x0003 on DSL 0. Layer 3 output may not apply
ISDN Serial1/0:23 interface
        dsl 0, interface ISDN Switchtype = primary-ni
        L2 Protocol = Q.921 0x0000  L3 Protocol(s) = CCM MANAGER 0x0003
    Layer 1 Status:
        ACTIVE
    Layer 2 Status:
        TEI = 0, Ces = 1, SAPI = 0, State = TEI_ASSIGNED

HQ#sh mgcp end       
Interface T1 1/0
             ENDPOINT-NAME    V-PORT     SIG-TYPE   ADMIN
             S1/ds1-0/1@HQ     1/0:23        none   up   
             S1/ds1-0/2@HQ     1/0:23        none   up   
             S1/ds1-0/3@HQ     1/0:23        none   up   
Interface T1 1/1
             ENDPOINT-NAME    V-PORT     SIG-TYPE   ADMIN
HQ#sh mgcp stat
  .........
 IP address based Call Agents statistics:
 IP address 10.1.200.21, Total msg rx 77,
                  successful 68, failed 9
      ................................ 
HQ#sh ccm-manager hosts 
MGCP Domain Name: HQ
Priority        Status                   Host
============================================================
Primary         Registered               10.1.200.21
First Backup    None                    
Second Backup   None                    

Call-manager-fallback config for SRST and non-SRST modes

For H323 GW interworking with CUCM cluster, User needs to config dial-peers in order to handle in/out calls from PSTN. For ex:

dial-pee voice 1 pots    ==== to handle incoming calls from PSTN
    incoming called .
    direct-inward
    port 1/0:23
dial-p voice 3120 voip    ==== to forward calls to CUCM
    destination-pat 1312301....$
    session target ipv4:10.1.200.21
    pref 1
dial-p v 900 pots
    desc ==== local  ====    note that this dial-peer does NOT have prefix 9 because CM strips digit 9
    destination-pat [2-9]......$
    port 1/0:23

In SRST mode, any local call with 7 digit would fail if call-manager-fallback is configured with secondary-dialtone 9

To get around this issue configure voice translation-rule to strip out digit 9 when dial-out of PSTN

voice translation-rule 100
        rule 1 /^9\(.*\)$/    /\1/        <<<  strip any string of digit beginning with 9
    voice translation-profile SRST-LOCAL
        translate called 100
call-manager-fallback
    translation-profile incoming SRST-LOCAL

Also, add translation-rule to trim incoming 10 digits to 4 digit
voice translation-rule 1
   rule 1 /^312301\(....\)$/   /\1/   ==== strip out all digits except last 4
voice translation-profile DID
   translate called 1
dial-peer voice 1 pots
   translation-profile incoming DID
   incoming called-number .
   direct-inward-dial
    port 1/0:23

In case SRST still did not work without adding digit 9 in the POTS dial-peer, add CLI  "access-code pri 9 direct-inward-dial" into call-manager-fallback config. Below is full config for call-manager-fallback:

voice translation-rule 1
 rule 1 /^1312301\(....\)/ /\1/
 rule 2 /^312301\(....\)/ /\1/
!
voice translation-profile DID
 translate called 1
!
application
 service alternate default
 !
 global
  service alternate default
 !
dial-peer voice 1 pots
 translation-profile incoming DID
 incoming called-number .
 direct-inward-dial
 port 1/0:23
!
dial-peer voice 911 pots
 description ==== 911
 destination-pattern 911
 port 1/0:23
 forward-digits 3
!
dial-peer voice 1000 pots
 description ==== local
 destination-pattern [2-9]......$
 port 1/0:23
!
dial-peer voice 1001 pots
 description ==== LD
 destination-pattern 1[2-9]..[2-9]......$
 port 1/0:23
 prefix 1
!
dial-peer voice 1002 pots
 description ==== INTL
 destination-pattern 011.T
 port 1/0:23
 prefix 011
!
dial-peer voice 2 voip
 description ==== to CUCM pub
 preference 1
 destination-pattern 312301....$
 session target ipv4:10.1.200.21
!
dial-peer voice 3 voip
 description ==== to CUCM sub
 preference 2
 destination-pattern 312301....$
 session target ipv4:10.1.200.20
!        
!
gateway
 timer receive-rtp 1200
!
!
call-manager-fallback
 secondary-dialtone 9
 max-conferences 12 gain -6
 transfer-system full-consult
 ip source-address 10.10.32.2 port 2000
 max-ephones 2
 max-dn 4
 system message primary SRST@BR1
 dialplan-pattern 1 3123012... extension-length 4
 transfer-pattern .T
 access-code pri 9 direct-inward-dial
 voicemail 917752011500
 call-forward pattern .T
 call-forward busy 917752011500
 call-forward noan 917752011500 timeout 10
!

Single Number Reach - aka SNR in 5 minutes

SNR - single number reach in 5 minutes

1. Config End-user hquser3, choose "Enable Mobility" and associate user with device (phone3)
2. On Menu device/device setting/softkey template, copy standard user template to Mobile user template. Click on "configure softkey layout" link. Add Mobility softkey to on-hook and connected states.
3. Config hqphone3 with Mobility softkey-template, and turn "Device Mobility Mode" ON
4. Add new device/device setting/remote destination profile - ex: hquser3.
        - userid = hquser3
        - DP = DP-HQ
        - CSS = CSS-global
5. Add new DN 1003 on remote destination profile ,
        - PT = PT-USA  (same PT as hqphone3/Line1)
    then save
6. Add Associated Remote Destinations
        - check both Mobile Phone & Enable Mobile Connect
        - destination number = 918005553333
7. Make sure Association Information - Line Association BOX is checked
8. Test by dial 1003, external phone 18005553333 should ring. Otherwise, reset  hqphone3 and recheck config

Install CUPC in 10 minutes

    1. on CM create a PresenceSIP trunk with DP-HQ and location.
    2. use same hquser1 (or hqphone1) for Presence, verify that hquser1 is associated to HQphone1 device and belong to Standard CTI enabled group. Also verify HQphone1 device "Owner User ID" is hquser1
    3. add Application User CtiGw with CTI enabled and control of all CTI devices
    4. verify System/Service Parameters/CUP Publish Trunk is set to PresenceSIP trunk name
    5. set CM license capability for users to CUP & CUPC
    6. add presence as an application server in CM
    7. Bring up Presence and add in CM info . Need CM security password
    8. Config CUP system/security/incoming & outgoing ACL to ALL
    9. set CUP system/service parameters  proxy-domain to CUPS.cisco.com
    10. add CM as a Presence gateway under menu Presence/Gateway
    11. configure routing at Presence/Routing to Default Cisco SIP Proxy TCP Listener
    12. set TFTP server under Application/CUPC/Settings with CM IP address and Default Cisco SIP Proxy TCP Listener
    13. under Application/CUPC/User Setting , select hquser1 and sign CTI profile to DP-HQ_cti_tcp_profile_synced_000.  If user hquser1 is not present, verify CM and CUPS sync in previous steps
    14. on Application/Desktop Control/Settings, set status to ON and fill in rest of info for CtiGw user
   
Verify with CUPC Desktop control on the local PC